Showing papers on "Microphone array published in 1990"

Switching adaptive filters for enhancing noisy and reverberant speech from microphone array recordings

Abstract: Switching adaptive filters, suitable for speech beamforming, with no prior knowledge about the speech source are presented. The filters have two sections, of which only one section at any given time is allowed to adapt its coefficients. The switch between both is controlled by a speech detection function. The first section implements an adaptive look direction and cues in on the desired speech. This section only adapts when speech is present. The second section acts as a multichannel adaptive noise canceller. The obtained noise references are typically very bad; hence, adaptation must be restricted to silence-only periods. Several ideas were explored for the first section. The most robust solution, and the one with the best sound quality, was given by the simplest solution, i.e. a delay and sum beamformer that cues in on the direct path only and neglects all multipath contributions. Tests were performed with a four-microphone array in a highly reverberant room with both music and fan type noise as jammers, SNR improvements of 10 dB were typical with no audible distortion. >

Noise reduction based on microphone array with LMS adaptive post-filtering

Abstract: This letter presents a self-adapting noise reduction system which is based on a 4-microphone array combined with an adaptive Wiener filter. The LMS algorithm is used for adaptation. This filtering structure allows a simple implementation in the time domain on a sample-by-sample basis.

Electro-acoustic system

Abstract: Electro-acoustic system for improving the acoustic of a predetermined room, said system comprising a microphone array having a plurality of microphones (2) and a loudspeaker array having a plurality of loudspeakers (6), as well as a signal processing unit (4), interposed between said arrays, said signal processing unit having means for generating reflections, whereby at least one of the microphones is directed in such a manner that it receives at least reflected sound from a sound source in the predetermined room and/or that at least one of the loudspeakers is directed at a reflecting surface in the predetermined room.

An algorithm for determining talker location using a linear microphone array and optimal hyperbolic fit

Abstract: One of the problems for all speech input is the necessity for the talker to be encumbered by a head-mounted, hand-held, or fixed position microphone. An intelligent, electronically-aimed unidirectional microphone would overcome this problem. Array techniques hold the best promise to bring such a system to practicality. The development of a robust algorithm to determine the location of a talker is a fundamental issue for a microphone-array system. Here, a two-step talker-location algorithm is introduced. Step 1 is a rather conventional filtered cross-correlation method; the cross-correlation between some pair of microphones is determined to high accuracy using a some-what novel, fast interpolation on the sampled data. Then, using the fact that the delays for a point source should fit a hyperbola, a best hyperbolic fit is obtained using nonlinear optimization. A method which fits the hyperbola directly to peak-picked delays is shown to be far less robust than an algorithm which fits the hyperbola in the cross-correlation space. An efficient, global nonlinear optimization technique, Stochastic region Contraction (SRC) is shown to yield highly accurate (>90%), and computationally efficient, results for a normal ambient.

Experimental results showing the effects of optimal spacing between elements of a linear microphone array

Abstract: Stochastic region contraction is applied to an improved field model which includes attenuation due to the source-to-microphone distance, and a new optimal spacing is derived. The performances of equispaced, logarithmically spaced, old-model optimal, and new-model optimal arrays were measured experimentally and are compared to the results predicted from the mathematical model. The shape of the power distribution of the received signal in the near field of the array is verified in general. The differences are due to echoes, background noise, nonideal microphone beam patterns, and other variables that were not included in the model. >

A microphone-array configuration for AMNOR

Abstract: This paper describes a microphone-array configuration for AMNOR (Adaptive Microphone-array system for NOise Reduction). AMNOR performance (S/N improvement) depends on the microphone-array configuration. Therefore, the chosen configuration is examined under three different sound field conditions by simulation experiment. The distance between two microphones is first described in detail to show that the optimum microphone distance is one-half the wavelength of the highest frequency in the broadbandnoise of interest. Based on these results, linear and circular-array configurations are next examined. These configurations can obtain similar results by the introduction of apparent microphone distance. The maximum value of this distance should also be one-half the wavelength of the highest frequency of the broadband noise.

Constrained Bptimum Filtering for Multi-Microphone Digital Hearing Aids

Abstract: An adaptive digital spatial filter is applied to a multiple microphone array which will be used as a hearing aid input. Models are employed to accurately reflect the effects of acoustic headshadow on the microphone array. Robust constraints are applied to a minimum variance cost function and simulated under a variety of optimum (non-adaptive) processing scenarios. Promising results for some of the methods evaluated are presented.

DSP beamforming and talker tracking with a linear microphone array

Abstract: One of the problems for all speech input is the necessity for the talker to be encumbered by a head‐mounted, hand‐held, or fixed position microphone. An intelligent, electronically aimed undirectional microphone would overcome this problem. Array techniques hold the best promise to bring such a system to practicality, although the acoustic problems are manifold. High‐speed digital signal processing (DSP) chips have made it possible to attack the essential problems of forming the beam, aiming the microphone, and tracking a unique talker. A useful beamforming algorithm and a two‐step talker‐tracking algorithm are introduced. In the latter, step 1 is a rather conventional filtered cross‐correlation method; the delay between some pair of microphones is determined to high accuracy using interpolation on the sampled data. Then, using the fact that the delays for a point source should fit a hyperbola, a best hyperbolic fit is obtained using nonlinear optimization. Results indicate that this method works reliably for signal‐to‐noise ratios of less than 10 dB. [This work principally supported by NSF Grant No. MIP‐8809742 and DARPA/NSF Grant No. IRI‐8901882.]

Evaluation of reverberant room impulse responses, including the effects of head shadow

Abstract: Simulation and analysis of multimicrophone hearing aids in realistic environments require accurate knowledge of room impluse responses at various points on or near the head. A two‐step method is presented that provides the room impulse responses for arbitrary locations around the head. The first step uses the method of images [J. B. Allen and D. B. Berkley, J. Acoust. Soc. Am. 65, 943–950 (1979)] to compute the impulse delay and attenuation from each of the various images. The second step factors incorporate the measured responses at various points on the head (or manikin) into the total room/head impulse response. The obtained responses can be used to process speech or other materials to simulate reverberant environments, and to generate array response vectors for multimicrophone hearing aid processors. These vectors can in turn be used to evaluate the sensitivity of the microphone array response to changes in the soundfield and to indicate the level of robustness required for hearing aid processors.