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Showing papers on "Microphone array published in 1987"


PatentDOI
TL;DR: In this article, a sound processing arrangement couples sound from a prescribed environment through a fixed microphone array to a signal processing arrangement having a specifiable preferred sound source location, where the microphone pickup signals are combined with a set of weighting signals to adjust the directional response pattern in successive analysis time intervals.
Abstract: A sound processing arrangement couples sound from a prescribed environment through a fixed microphone array to a signal processing arrangement having a specifiable preferred sound source location. The microphone pickup signals are combined with a set of weighting signals to adjust the directional response pattern in successive analysis time intervals. The weighting signals are modified in each analysis time interval so that the total acoustic signal power of the signal processing arrangement output signal is decreased toward a minimum while substantially unity power transfer of sound signals from said preferred location is maintained at all frequencies over a prescribed frequency range. In this way, the preferred source location is in the main beam while unwanted sound source locations are at the null points of the adjusted directional response pattern.

99 citations


Journal Article
TL;DR: An adaptive beamforming method that functions to preserve target signals arriving from straight-ahead of a microphone array while minimizing output power from off-axis interference sources is described.
Abstract: To reduce interference in monaural hearing aids from sound sources that are spatially separated from a target source, we are investigating methods for combining information from multiple microphones. In this paper, we describe an adaptive beamforming method that functions to preserve target signals arriving from straight-ahead of a microphone array while minimizing output power from off-axis interference sources. In a preliminary evaluation of a two-microphone system, sentence intelligibility tests were administered to normal-hearing subjects using processed and unprocessed materials from simulated environments in which the target was on-axis, the interference (speech babble) was 45 degrees off-axis, and the reverberation mimicked that of a living room, a conference room, and anechoic space. Compared to listening through a single microphone, the two-microphone beamformer reduced the target-to-interference ratio required to achieve 50 percent keyword intelligibility by 30, 14, and 0 dB in the anechoic, living-room, and conference-room conditions, respectively. The corresponding improvements over binaural listening (one microphone to each ear) were 24, 9, and 0 dB. Further tests in the living-room environment using the same beam-forming system but with filter impulse responses shortened by a factor of four (which would decrease the adaptation time by a factor of four) decreased the improvement by 5 dB. These results are sufficiently encouraging to warrant further tests involving more realistic reverberant conditions, multiple sources of interference, and time-varying acoustic environments.

72 citations


Journal ArticleDOI
TL;DR: Several sets of curves are presented from the solution of constrained and unconstrained multidimensional nonlinear equations derived from the theory and suggest optimal spacing and gains for linear microphone arrays for speech acquisition.
Abstract: Speech data for recognition, talker verification, or recording are typically acquired using a head-mounted or hand-held microphone. These devices can be very inconvenient or provide a poor signal-to-noise ratio. A microphone array has the potential for surmounting both of these problems. Here, results on optimal spacing and gain for practical linear arrays are derived under the assumption that the desired signal and intrusive speech may be accurately modeled by plane waves. Several sets of curves are presented from the solution of constrained and unconstrained multidimensional nonlinear equations derived from the theory. The curves suggest optimal spacing and gains for linear microphone arrays for speech acquisition.

66 citations


19 May 1987
TL;DR: In this paper, an acoustic microphone array for sound measurements outdoors, with applications in industrial noise and traffic noise, was designed and developed, with a flexible length of 10 to 76 m and covers the octave band from 125 to 1000 Hz (later extended to 2000 Hz).
Abstract: Design and development of an acoustic microphone array for sound measurements outdoors, with applications in industrial noise and traffic noise. The microphone array has a flexible length of 10 to 76 m and covers the octave band from 125 to 1000 Hz (later extended to 2000 Hz). The angular resolution is 1.5 degrees.

7 citations


Patent
08 Apr 1987
TL;DR: In this paper, a ring of outwardly facing microphones is used to respond to sound from a target in a range-finder system, which is called a sonic rangefinder (RSF).
Abstract: A sonic range-finder uses as its responsive elements a ring of outwardly-facing microphones (a, b, c), which respond to sound from a target (1). If the range-finder is active it has a transmitter (2), which emits ultrasonic pulses at, for instance, one per half-second, in broadcast manner. The pulses reflected from the target are detected by two or more microphones and the relative responses indicate the range and direction of the target. If the range-finder is passive, operation is the same, except that it relies on sound from the target. The microphone array is a small (e.g. 60 cm in diameter) circular array of outwardly facing electret microphones stretched onto a printed circuit band and secured to a cylindrical member.

1 citations


Journal ArticleDOI
TL;DR: This system uses an array of sensors, e.g. microphones, which gives the input to digital and adaptive FIR filters which cancel unwanted noise which comes from a few unknown directions and still maintain constant gain in a known look direction.