Chair: Schuyler Quackenbush, AT&T Bell Laboratories (USA)
Richard Leahy, University of Southern California (USA)
Zhenyu Zhou, University of Southern California (USA)
Yung-Chih Hsu, University of Southern California (USA)
We describe a new class of algorithms for active noise control (ANC) for use in environments in which impulsive noise is present. The well known filtered-X and filtered-U algorithms ANC algorithms are designed to minimize the variance of a measured error signal. For impulsive noise, which can be modeled using non-Gaussian stable processes, these standard approaches are not appropriate since the second order moments do not exist. We propose a new class of adaptive algorithms for ANC that are based on the minimization of a fractional lower order moment, p < 2. By studying the effect of p on the convergence behavior of adaptive algorithms, we observe that superior performance is obtained by choosing p approximately equal to alpha where alpha < 2 is a parameter reflecting the degree of impulsiveness of the noise. Applications of this approach to noise cancellation in a duct are presented.
Gang-Janp Lin, National Chia Tung University (REPUBLIC OF CHINA)
Sau-Gee Chen, National Chia Tung University (REPUBLIC OF CHINA)
Terry Wu, National Chia Tung University (REPUBLIC OF CHINA)
A high-quality and low-complexity algorithm for pitch modification of acoustic signals is proposed. It is high quality, because time-domain waveform shape and phase syn chrony of a synthesized signal closely resemble that of its original signal. Low complex ity is at tained by performing the algorithm en tirely in time domain with fast algorithm, and without resorting to complicated frequency do main analy sis and synthesis. The time domain synthesis mostly consists of fast algorithms of finding minimum absolute error (MAE) and a cross fading operation for high correlation gain and phase synchrony. The MAE- based algorithm is shown to yield smaller complexity, and better performance, than other well-known cor relation cost functions. Pitch modification is performed by selecting and synthesizing apposite frames from original input signals in a way such that the origi nal waveform shape and phase synchrony are preserved. Subjective tests showed a comparable performance to that of the best known algorithms, but at much reduced complexity.
J. C. Tejero, F. Informatica Pl El Ejido S/N
S. Bernal, University of Malaga (SPAIN)
J. A. Hidaldo, University of Malaga (SPAIN)
J. Fernandez, University of Malaga (SPAIN)
R. Urquiza, University of Malaga (SPAIN)
A. Gago, University of Malaga (SPAIN)
We introduce a digital hearing aid that compensate the signal spoken in sensorineural impaired listeners with object of improving their intelligibility. The technique used is based on a digital analysis/synthesis of speech, we divided the input signal into short time blocks then we make a multiband analysis, non linear amplification and synthesis basing us in a sinusoidal model of the voice, according to the subjet's dynamic range in that band. This system has been implemented in real-time using a DSP (TMS320C30) based microprocessor board within a host personal computer IBM PC, having been one of the principal objectives of the project to make an easy implementation VLSI system and low consumption for it to be a portable digital hearing aid
Thomas Chou, Massachusetts Institute of Technology (USA)
A method is developed for designing broadband beamformers with highly frequency-invariant behavior. The method combines harmonic nesting with filter-and-sum beamforming, yielding systems with efficient log-period-ic transducer structures and plane wave responses which deviate from the desired response by less than 1% of the peak mainlobe response. In particular, null locations are very constant with frequency, allowing the nulling of directional broadband interference. A digital implementation covering audio frequencies (500 - 7200 Hz) is built and demonstrates results in good agreement with simulations, yielding deviations under 5% of peak response.
Peter L. Chu, PictureTel Corporation (USA)
Reducing the noise and reverberance in sound pickup has been a problem ever since the microphone was invented. Elegant solutions using multiple microphones in an array are a current hotbed of research. Unfortunately, because of the computational/monetary cost of these approaches, they have not been widely implemented in products. In this paper, an automatically steered mic array, which works by taking linear combinations of two dipole microphones, is presented whose cost is low enough to have been implemented in a videoconferencing product. The array is positioned centrally on the conference table and provides very reasonable pickup for people speaking within a 7 foot radius, adequate for most conferencing situations. While simple in structure, the array provides a large increase in convenience and performance compared to the common method of laying out multiple cardioid microphones on the table, where each participant must be within the pickup angle/range of a cardioid microphone.
Nathaniel A. Whitmal, Northwestern University
Janet C. Rutledge, Northwestern University
Jonathan Cohen, DePaul University (USA)
A novel method for enhancement of noisy speech is presented. Frames of speech samples are split into low and high frequency bands and projected onto a library of bases consisting of local trigonometric functions and wavelet packets. Coefficients thought to represent only the speech are selected by means of the MDL criterion, and used to synthesize an estimate of the original speech. A tracking algorithm uses MDL values to choose between the MDL processor and alternate processors which reject audible artifacts. Preliminary results indicate that the new algorithm may be useful in applications requiring a single-microphone noise reduction system for speech.
Hong Wang, PictureTel Corporation (USA)
A new approach of estimating acoustic transfer functions from multi-channel reverberant signals is proposed. The ratio of two transfer functions is approximated by a polynomial by estimating one reverberant signal from another using least mean square estimation. Each transfer function is then estimated using Pade approximation. Two criteria are used to search for the most appropriate pair of transfer functions among all pairs generated by Pade approximation. The dereverberated signal is derived by multi-channel inverse filtering using the multiple input/output inverse theorem. Simulation using Gaussian noise convolved with minimum phase transfer functions gave a reconstruction SNR of 66dB. The approach is also applied to sub-band inverse filtering of reverberant speech recorded in a real room.
C. Rogers, East Texas State University (USA)
Bird songs and calls have been studied using modern signal processing techniques in both time and frequency domains. The complex modulations of these natural sounds are interpreted by a popular bioacoustic model of avian sound production. High resolution signal processing techniques have been applied to the problem of dual-voice detection within birdsongs. Improved time and frequency analyses of these signals have been achieved.
Stuart E. Kirtman, The Cooper Union for the Advancement of Science and Art
Harvey F. Silverman, Brown University (USA)
This paper describes the design and use of a digital acoustic array signal processing research environment. The system unites the interactive numerical processing capabilities of The MathWorks' MATLAB with multichannel data acquisition / signal processing hardware and an overhead Cartesian robot for accurate sound source placement. In conjunction with additional electronic and electromechanical peripherals, the user-friendly environment facilitates the spatial and temporal analysis and evaluation of acoustic array algorithms. Real-data experiments and their results are provided to demonstrate the flexibility and ease-of-use of the array system.
Michael S. Brandstein, Brown University (USA)
John E. Adcock, Brown University (USA)
Harvey F. Silverman, Brown University (USA)
The linear intersection (LI) estimator, a closed-form method for the localization of source positions given only the sensor array time-delay estimate information, is presented. The array is constrained to be composed of 4-element sub-arrays configured in 2 centered orthogonal pairs. A bearing line in 3-space is estimated from each sub-array and potential source locations are found via closest intersection of bearing line pairs. The final location estimate is determined by a probabilistic weighting of these potential locations. The LI estimator is shown to be robust and accurate, to closely model the ML estimator, and to outperform a representative algorithm. The computational complexity of the LI estimator is suitable for use in real-time microphone-array applications.