Showing papers on "Microphone array published in 1992"

Hands-free voice communication in an automobile with a microphone array

Abstract: The authors present the result of their research on developing a hands-free voice communication system with a microphone array for use in an automobile environment. The goal of this research is to develop a speech acquisition and enhancement system so that a speech recognizer can reliably be used inside a noise automobile environment, for digital cellular phone application. Speech data have been collected using a microphone array and a digital audio tape (DAT) recorder inside a real car for several idling and driving conditions, and processed using delay-and-sum and adaptive beamforming algorithms. Performance criteria including signal-to-noise ratio and speech recognition error rate have been evaluated for the processed data. Detailed performance results presented show that the microphone array is superior to a single microphone. >

A microphone array for car environments

Abstract: The authors describe a microphone array for speech recording in car environments. The array is designed for hands-free radiotelephone, and is also used as a front-end for an automatic speech recognition system. The configuration of the array and the adaptive beamforming technique implemented are described and the performance of the array are evaluated. The measure of performance is the score obtained with a speech recognition system. Within the European ESPRIT project ARS (adverse environment recognition of speech), a prototype of this microphone array was built. It has eight microphones and works in real time, using one TMS C30 processor. >

A wideband blind identification approach to speech acquisition using a microphone array

Abstract: The problem of speech separation from an array of microphones is considered. The problem is modeled as a wideband process with unknown model parameters, namely, sensor gains and time delays. The technique of blind identification is extended to the wideband case. It is shown that the wideband blind identification approach is capable of handling multipath situations. Simulations are presented which show that the wideband signals can indeed be separated by the proposed approach. >

Experimental results for baseline speech recognition performance using input acquired from a linear microphone array

Abstract: In this paper, baseline speech recognition performance is determined both for a single remote microphone and for a signal derived from a delay-and-sum beamformer using an eight-microphone linear array. An HMM-based, connected-speech, 38-word vocabulary (alphabet, digits, 'space', 'period'), talker-independent speech recognition system is used for testing performance. Normal performance, with no language model, i.e., raw word-level performance, is currently about 81% for a set of talkers not in the training set and about 91% for training set data. The system has been trained and tested using a close-talking head-mounted microphone. Since a meaningful comparison requires using the same speech, the existing speech database was appropriately pre-filtered, played out through a transducer (speaker) in the room environment, picked-up by the microphone array, and re-stored as a digital file. The resulting file was post-processed and used as input to the recognizer; the recognition performance indicates the effect of the input device. The baseline experiment showed that both a single remote microphone and the beamformed signal reduced performance by 12% in a room with no other talkers. For the array tested, the error is generally attributable to reverberation off the floor and ceiling.

Voice signal input system

Abstract: PURPOSE:To emphasize only a voice signal generated from a speaker and to suppress other unnecessary noise signals by arranging respective microphones in a two-dimensional microphone array so that speaker's voice can be collected at maximum. CONSTITUTION:This voice signal input system is provided with the two-dimensional microphone array 1 arranged so as to collect a voice input signal at maximum from a virtual point sound source and a mixer circuit 2 for adding input signals from the array 1. When a sound wave arrives at the array 1, three signals inputted to the mixer 2 are emphasized by the addition of a same phase signal and the emphasized signal is outputted to a microphone amplifier part 3. When the three signals having respectively different phases are inputted to the mixer 2 and added, the added signal whose amplitude is reduced is outputted to the amplifier part 3. Since only a necessary speaker's voice can be selectively emphasized, other unnecessary noise components can be suppressed.

Sound surveillance equipment and method

Abstract: PURPOSE:To make the search of a sound source in water, etc., easy with simple constitution. CONSTITUTION:Signals of sounds extracted with several microphones 1a, 1b,..., 1k constituting a microphone array 1 are individually made phase modulation through delay units 2a, 2b,..., 2k, and the specification of sound source position is made possible by way of making the result of adding outputs of several delay units 2a, 2b,..., 2k with an adder 3 maximum if a real sound source is in the estimated position of sound source. The said estimated sound source position is scanned and the real sound source is detected.

Near‐field array with constant focal area.

Abstract: A microphone array to be placed in the headliner of an automobile was designed for Cellular telephone application. The design goal was to provide a focal area of 20×20 cm centered at the driver’s mouth to be held constant from 300 to 3000 Hz. Digital filtering‐delay techniques were used to control focal area. Array size limitations and a limited number of elements produced focal area sizes of 19×19 to 24×24 cm from 450 to 3000 Hz.

Classical Recording and Production

Abstract: This chapter will discuss the musical and technical factors involved in producing classical recordings for commercial release. It will be concerned with selecting the recording venue, planning the sessions, placing the microphones in order to produce a desired recording perspective, and planning logistical details of equipment and staffing.

Recording the Spoken Voice

Abstract: In this chapter we will discuss the problems of recording the voice, both for narration and for drama, with emphasis on microphone choice and placement, local acoustical considerations, signal processing, and monitoring

Two-Channel Stereo

Abstract: Stereophonic sound, or stereo as it is usually called, refers to any system of recording or sound transmission using multiple microphones and loudspeakers. Signals picked up by the microphones are routed to loudspeakers that are placed in a geometrical array corresponding to the microphone array. In this manner, many of the spatial aspects of the recording environment are preserved, and the listener can perceive, more or less accurately, the spatial perspective of the original performance in its acoustical surroundings. Stereo need not be limited to two channels. Motion picture systems have included upwards of six channels. For home use, however, stereo is presently limited to two transmission channels.