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Jens Meyer, Darmstadt University of Technology (Germany)
Carsten Sydow, SIEMENS AG (Germany)
In this paper an application of the noise cancelling method for suppression of noise of a microphone array system is discussed. First an overview of the noise cancelling approach is given. This is followed by a description of the employment of the method in a realized microphone array system. The limiting factors are described and theoretical limits of the noise suppression are derived. Experimental results, which are obtained in a realistic environment, are presented. The results show, that depending on the recording situation the noise cancelling approach applied to a microphone array system leads to a significant enhancement of the signal to noise ratio of the array output signal.
ic970211.pdf ic970211.pdf
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Kenji Kiyohara, NTT Human Interface Labs. (Japan)
Yutaka Kaneda, NTT Human Interface Labs. (Japan)
Satoshi Takahashi, NTT Human Interface Labs. (Japan)
Hiroaki Nomura, NTT Human Interface Labs. (Japan)
Junji Kojima, NTT Human Interface Labs. (Japan)
This paper proposes a microphone array system which realizes the following important functions for speech recognition: i) SNR improvement, ii) flat spectrum response for an arbitrary speaker position, and iii) speech period detection in noisy speech. This microphone array system features time delay estimation using pre-whitening signal processing, delay-and-sum array weighted optimally, and speech period detection based on the level difference (called MLD) between signals before and after array processing. Word recognition experiments performed in the presence of crowd noise demonstrate greater robustness of the proposed system against noise than the system with conventional directional microphone and speech period detection method.
ic970215.pdf TOPWalter Kellermann, FH Regensburg (Germany)
New concepts for efficient combination of acoustic echo cancellation (AEC) and adaptive beamforming microphone arrays (ABMA) are presented. By decomposing common beamforming methods into a time-invariant part, which the AEC can integrate, and a separate time-variant part, the number of echo cancellers is minimized without rendering the system identification problem more difficult. Methods for controlling the interaction of ABMA and AEC are outlined and implementations for typical microphone array applications are discussed briefly.
ic970219.pdf ic970219.pdf TOP
Gary W. Elko, Acoustics Research Department (U.S.A.)
Anh-Tho Nguyen Pong, Speech Processing Software and Technology Research (U.S.A.)
A new first-order differential microphone array with an infinitely steerable and variable beampattern is described. The microphone consists of 6 small pressure microphones flush-mounted on the surface of a 3/4" diameter rigid nylon sphere. The microphones are located on the surface at points where included octahedron vertices contact the spherical surface. By appropriately combining the three Cartesian orthogonal pairs with simple scalar weightings, a general first-order differential microphone beam (or beams) can be realized and directed to any angle in 4(pi) steradian space. A working real-time version has been created and measured results from this microphone are shown. This microphone should be useful for surround sound recording/playback applications and to virtual reality audio applications.
ic970223.pdf ic970223.pdf TOP
Maurizio Omologo, ITC-IRST (Italy)
Marco Matassoni, ITC-IRST (Italy)
Piergiorgio Svaizer, ITC-IRST (Italy)
Diego Giuliani, ITC-IRST (Italy)
The use of a microphone array for hands-free continuous speech recognition in noisy and reverberant environment is investigated. An array of eight omnidirectional microphones was placed at different angles and distances from the talker. A time delay compensation module was used to provide a beamformed signal as input to a Hidden Markov Model (HMM) based recognizer. A phone HMM adaptation, based on a small amount of phonetically rich sentences, further improved the recognition rate obtained by applying only beamforming. These results were confirmed both by experiments conducted in a noisy and reverberant environment and by simulations. In the latter case, different conditions were recreated by using the image method to reproduce synthetic versions of the array microphone signals.
ic970227.pdf ic970227.pdf TOP
Piergiorgio Svaizer, ITC-IRST (Italy)
Marco Matassoni, ITC-IRST (Italy)
Maurizio Omologo, ITC-IRST (Italy)
A microphone array can be used to locate a dominant acoustic source in a given environment. This capability is successfully employed to locate an active talker in teleconferencing or other multi-speaker applications. In this work the source location is obtained in two steps: 1) a Time Difference Of Arrival (TDOA) computation between the signals of the array; 2) an ``optimal'' source location based on the interchannel delay estimates and on a geometrical description of the sensor arrangement. The Crosspower Spectrum Phase technique was used for TDOA estimation, while a Maximum Likelihood approach was followed to derive the source coordinates. Source location experiments in a three-dimensional space were performed by means of an array of 8 microphones. For this purpose both a loudspeaker and a real talker were used to collect data in a large noisy and reverberant room.
ic970231.pdf ic970231.pdf TOPPeter Chu, PictureTel (U.S.A.)
In set-top videoconferencing, the complete videoconferencing system fits unobtrusively on top of the television. The microphone sound pickup system is one of the most important functional blocks with constraints of small size, high performance, and low cost. Persons speaking several feet away from the system must be picked up satisfactorily while noise generated internally in the system by the cooling fan and hard drive, and noise generated externally from air conditioning and nearby computers must be attenuated. In this paper, a three microphone superdirective array is described which meets these constraints. An analog highpass and lowpass filter are used to merge two of the microphone signals to form a single channel, so that a single stereo A/D converter is required to process the three microphone signals. The microphone signals are then linearly combined so as to maximize the signal-to-noise ratio, resulting in nulls steered toward nearby objectionable noise sources.
ic970235.pdf ic970235.pdf
3121_a.wav 3 microphone array, about 4 ft. from source to array TOP
Matttias Dahl, University of Karlskrona/Ronneby (Sweden)
Ingvar Claesson, University of Karlskrona/Ronneby (Sweden)
Sven Nordebo, University of Karlskrona/Ronneby (Sweden)
This paper presents a method to simultaneously perform 20~dB acoustic echo cancellation and 15-20~dB speech enhancement using an adaptive microphone array combined with spectral subtraction. Primarily intended for handsfree telephones in automobiles, the microphone array system simultaneously emphasizes the near-end talker and suppresses the handsfree loudspeaker and the broadband car noise. The array system is based on a fast and efficient on-site calibration and can be used in other situations such as conventional speaker phones.
ic970239.pdf ic970239.pdf 3122_a.wav Microphone Ericsson: Echo and noise cancelling, using one microphone (0-8~sec) vs. using an on-site calibrated echo and noise cancelling array with two (8-16~sec) and six microphones (16-24~sec). 3122_b.wav Microphone Ericsson: Echo and noise cancelling, using one microphone (0-8~sec) vs. using an on-site calibrated echo and noise cancelling array with two (8-16~sec) and six microphones (16-24~sec). The subsequent spectral subtraction system is present. 3122_c.wav Microphone Sennheiser: Echo and noise cancelling, using one microphone (0-8~sec) vs. using an on-site calibrated echo and noise cancelling array with two (8-16~sec) and six microphones (16-24~sec). The two figures illustrates how the system works with and without the subsequent spectral subtraction system. 3122_d.wav Microphone Sennheiser: Echo and noise cancelling, using one microphone (0-8~sec) vs. using an on-site calibrated echo and noise cancelling array with two (8-16~sec) and six microphones (16-24~sec). The subsequent spectral subtraction system is present. TOP
Sven Nordholm, University of Karlskrona/Ronneby (Sweden)
Ingvar Claesson, University of Karlskrona/Ronneby (Sweden)
This paper gives an analytical description of an adaptive microphone array which facilitates a simple built-in calibration to the environment and instrumentation. The scheme offers several advantages, such as a simple calibration procedure and reduced target signal distortion. The analysis employs noncausal Wiener filters yielding compact and effective theoretical suppression limits.
ic970243.pdf ic970243.pdf TOP
Yves Grenier, ENST (France)
Sofiène Affes, INRS-Télécommunications (Canada)
Matched filtering and adaptive beamforming are both necessary for efficient speech dereverberation and noise reduction by microphone arrays. This can be achieved by the identification of impulse responses. In this contribution, we show that adaptive microphone arrays are sensitive to identification errors of impulse responses, particularly due to speaker movements. We prove that adjusted matched-filtering and permanent tracking of impulse responses are also necessary. The proposed microphone array responds well to these requirements under realistic conditions.
ic970247.pdf ic970247.pdf TOP
Harvey F. Silverman, Brown University (U.S.A.)
William R. Patterson, Brown University (U.S.A.)
James L. Flanagan, Rutgers University (U.S.A.)
Daniel Rabinkin, Rutgers University (U.S.A.)
The Huge Microphone Array(HMA) project started in February 1994 to design, construct, and test a real-time 512-microphone array system and to develop algorithms for use on it. Analysis of known algorithms showed that signal-processing performance of over 6 Gigaflops would be required; at the same time, there was a need for portability, i.e., fitting into a small van. These tradeoffs and many others have led to a unique design in both hardware and software. This paper presents the design and its justifications. Performance data for a few important algorithms relative to usage of processing-capability, response latency, and difficulty of programming are discussed.
ic970251.pdf ic970251.pdf TOP