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Speech enhancement using microphone array with multi-stage processing

IEICE TRANSACTIONS ON FUNDAMENTALS OF ELECTRONICS COMMUNICATIONS AND computer SCIENCES 1996年第3期E79A卷 386-394页

作者： Cao, YC Sridharan, S Moody, M Speech Research Lab. Signal Processing Research Centre School of Electrical and Electronic Systems Engineering Queensland University of Technology Brisbane QLD QLD4001 GPO BOX 2434 Australia

A microphone array system with multi-stage processing for speech enhancement is presented in this paper. Two beamformers with uniform directional patterns, one aimed at the target source and the other at the interfering sources, convert the multi-channel inputs into two data sequences. A novel microphone array structure with a small aperture has been designed to obtain the dual beamformers. The outputs of the two beamformers are then presented to a post-processing stage to further improve the quality and intelligibility of the speech signal. The post-processing stage can be selected from one of three different algorithms that are presented, which are suitable for different acoustic environments. Applications for such a system include hands-free telephony, teleconferencing and also special situations where speech signals must be picked up in an extremely noisy acoustic environment in which the microphones are hidden (e.g. in a forensic covert recording system).

关键词： acoustics signal processing speech enhancement

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OPTIMAL LINEAR FILTERING FOR POWER ESTIMATION OF SLOWLY-VARYING COMPLEX-VALUED signalS

OPTIMAL LINEAR FILTERING FOR POWER ESTIMATION OF SLOWLY-VARY...

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1996 3rd International Conference on signal processing(ICSP’96)

作者： Aiping Huang Timo I.Laakso Seppo J.Ovaska Iiro O.Hartimo Lab.of Signal Processing & Computer Technology Helsinki Univ.of Technology Lab.of Telecommunications Technology Helsinki Univ.of Technology

A novel method for designing the FIR filter in the Hammerstein model for power estimation of complex-valued signals is *** enables a computationally effective optimal power estimation with a prescribed delay in minimum mean squared error sense,a great reduction of the estimation bias which is inherent in power *** are given by assuming a constant input signal and a Rayleigh distributed input fading signal respectively,both signals are complex-valued and corrupted by Gaussian noise.

关键词： MSE OPTIMAL LINEAR FILTERING FOR POWER ESTIMATION OF SLOWLY-VARYING COMPLEX-VALUED signalS POWER

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The rational behavior software architecture for intelligent ships

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NAVAL ENGINEERS JOURNAL 1996年第2期108卷 43-55页

作者： Byrnes, RB Healey, AJ McGhee, RB Nelson, ML Kwak, SH Brutzman, DP [?]Ronald B. Byrnes is a U.S. Army Lieutenant Colonel commissioned into the Signal Corps in 1979. He has served in a variety of positions associated with tactical and operational communications at company brigade and corps level. L. C. Byrnes became a member of the Army Acquisition Corps in 1993 and is currently a computer scientist with the Software Technology Branch of the Army Research Lab. His current work includes continuation of his doctoral research into high-level control of autonomous vehicles contributions to the Army's electronic records management and data standardization efforts and automated support for flight plan verification. L.C. Byrnes received a B.S. degree in Mathematics from Midwestern State University in 1979 an M.S.E.E. degree from the Naval Postgraduate School in 1989 and a Ph.D. in Computer Science from the Naval Postgraduate School in 1993.

The solutions to the power projection, transportation, and operational needs of the Navy as it faces the 21st century must account for reduced manning levels. This leads naturally to increased use of computers, automation, and intelligent systems in the concept and design of the next generation of ships. In addition to the acknowledged hardware needs, the problem of autonomic and autonomous control of shipboard systems and missions are amenable to and will, in fact, require software solutions. Despite current technology, large, reliable software systems are difficult to achieve because correctness in requirements analysis, design, implementation, testing, modification, and maintenance of software are difficult. Software is also difficult to quantize and display;hence, the effort and costs involved in its development are easily underestimated. This paper describes an approach to the problem of providing structure, in the form of a software architecture, to the software performing autonomous control of missions and their related tasks. In concert with the need to reduce complexity, the architecture must support simple, rapid reconfiguration of code should vehicle capabilities or mission requirements change. Building upon recent efforts with control of Autonomous Underwater Vehicles (AUVs), we propose a tri-level control system architecture called the Rational Behavior Model (RBM) as an approach to autonomous and autonomic control of surface ship missions and systems.

关键词： Electronic ship equipment

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FAST ALGORITHMS FOR ANALYZING AND DESIGNING WEIGHTED MEDIAN FILTERS

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signal processing 1995年第2期41卷 135-152页

作者： YANG, RK GABBOUJ, M NEUVO, Y [a] Audio-Visual Signal Processing Lab. Nokia Research Center Kanslerink 8N SF-33720 Tampere Finland [b] Signal Processing Laboratory Tampere University of Technology Tampere Finland [c] Nokia Mobile Phone Espoo Finland

In this paper, two fast algorithms are developed to compute a set of parameters, called M(i)'s, of weighted median filters for integer weights and real weights, respectively. The M(i)'s, which characterize the statistical properties of weighted median filters and are the critical parameters in designing optimal weighted median filters, are defined as the cardinality of the positive subsets of weighted median filters. The first algorithm, which is for integer weights, is abo ut four times faster than the existing algorithm. The second algorithm, which applies for real weights, reduces the computational complexity significantly for many applications where the symmetric weight structures are assumed. Applications of these new algorithms include design of optimal weighted filters, computations of the output distributions, the output moments, and the rank selection probabilities, and evaluation of noise attenuation for weighted median filters.

关键词： WEIGHTED MEDIAN FILTER OPTIMAL FILTERING IMAGE processing STATISTICAL PROPERTIES

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SPEECH ENHANCEMENT BY EIGEN DECOMPOSITION WITH TWO-CHANNEL OBSERVATIONS 4

SPEECH ENHANCEMENT BY EIGEN DECOMPOSITION WITH TWO-CHANNEL O...

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4th European Conference on Speech Communication and technology, EUROSPEECH 1995

作者： Cao, Yuchang Sridharan, Sridha Moody, Miles Speech Research Lab. Signal Processing Research Centre School of Electrical and Electronic Systems Engineering Queensland University of Technology GPO Box 2434 BrisbaneQ4001 Australia

The concept of eigen decomposition for two-channel signal separation, a novel method of enhancing the desired signal corrupted by interference, is described. The method uses two observations, which come from a pair of single sensors (or beamformers) both of which contain the desired signal and the undesired signal(s). The method assumes that the desired signal and the undesired signal(s) are uncorrelated and the signal-to-noise ratios (the ratio of the desired signal to the undesired signal(s)) of each observation are different. A new algorithm based on eigen analysis has been developed to eliminate the noise components sequentially. The technique has been successfully used to separate a speech signal corrupted heavily by ambient noise, co-talker interference and background music. © 1995 4th European Conference on Speech Communication and technology, EUROSPEECH 1995. All rights reserved.

关键词： Speech enhancement

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SPEECH-SEEKING MICROPHONE ARRAY WITH MULTI-STAGE processing 4

SPEECH-SEEKING MICROPHONE ARRAY WITH MULTI-STAGE PROCESSING

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4th European Conference on Speech Communication and technology, EUROSPEECH 1995

This paper describes an auto-seeking microphone array consisting of novel multiple processing stages for speech acquisition. The array automatically detects the speech in the presence of noise and tracks the speech sources. A combination of dual-beamforming and post-beamformer enhancement significantly im-proves the quality and intelligibility of desired speech at the output of the system. Applications for such a system include hands-free telephony, teleconferencing and special situations where speech signals must be picked up in a noisy acoustic environment and the microphones are hidden (as, e.g., in a forensic covert recording system). © 1995 4th European Conference on Speech Communication and technology, EUROSPEECH 1995. All rights reserved.

关键词： Beamforming

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A processor core for 32 kbit/s G.726 ADPCM codecs

A processor core for 32 kbit/s G.726 ADPCM codecs

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IEEE International Symposium on Circuits and Systems (ISCAS)

作者： J. Vehvilainen J. Nurmi Signal Process. Lab. Tampere Univ. of Technol. Finland Signal Processing Laboratory Tampere University of Technology Tampere Finland

This paper describes an application specific DSP core designed to be used in a CCITT 32 kbit/s G.726 Adaptive Differential Pulse Code Modulation (ADPCM) codec. The instruction set architecture and the programming model of the DSP core were derived from an algorithm profile and complexity analysis and the core was implemented using VHDL and logic synthesis. Architecture design efforts were concentrated on finding the minimum amount of hardware resources which could implement the required functionality within the clock cycle count limit. The result is a Harvard architecture processor core which can be used to implement the 32 kbit/s G.726 ADPCM encoding/decoding functions with very modest external instruction and data memory requirements. In a typical configuration the processor can perform a full encode/decode operation for one sample in less than 1100 clock cycles. A gate-level implementation of less than 4000 gates of silicon area was created using logic synthesis for a standard cell technology.

关键词： Codecs Digital signal processing Pulse modulation Logic programming Clocks Decoding Modulation coding Algorithm design and analysis Hardware Encoding

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On the characterization of impulsive noise with /spl alpha/-stable distributions using Fourier techniques

On the characterization of impulsive noise with /spl alpha/-...

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Asilomar Conference on signals, Systems & computers

作者： S.M. Koyon D.B. Williams Digital Signal Process. Lab. Georgia Inst. of Technol. Atlanta GA USA Digital Signal Processing Laboratory School of Electrical and Computer Engineering Georgia Institute of Technology Atlanta GA USA

A new technique is presented for the characterization of impulsive interference with /spl alpha/-stable processes. The proposed model is constructed by estimating the /spl alpha/-stable distribution parameters using the empirical characteristic function. Characteristic function based methods have been shown to yield the best performance but are plagued by the amount of computation required. The new parameter estimation algorithm is shown to achieve the same level of performance as other characteristic function based methods while greatly reducing the amount of computation required.

关键词： Parameter estimation Distributed computing Table lookup Digital signal processing lab.ratories Interference signal processing algorithms Radar Filtering Limiting

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Motion estimation and segmentation by fuzzy clustering

Motion estimation and segmentation by fuzzy clustering

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IEEE International Conference on Image processing

作者： B. Due P. Schroeter J. Bigun Signal Process. Lab. Swiss Federal Inst. of Technol. Lausanne Switzerland Signal Processing Laboratory Swiss Federal Institute of Technology Lausanne Switzerland

A layered motion estimation scheme using fuzzy clustering is introduced in this paper. Once motion estimation is performed, a lab.l smoothing is applied by using a modified objective function that takes into account s... 详细信息

关键词： Motion estimation Robustness Image sequences Optical scattering Image motion analysis Optical computing signal processing lab.ratories Smoothing methods Motion compensation

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Implementation of fractional delay waveguide models using allpass filters

Implementation of fractional delay waveguide models using al...

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International Conference on Acoustics, Speech, and signal processing (ICASSP)

作者： V. Valimaki M. Karjalainen Lab. of Acoust. & Audio Signal Process. Helsinki Univ. of Technol. Espoo Finland Laboratory of Acoustics and Audio Signal Processing Helsinki University of Technology Espoo Finland Center for Computer Research in Music and Acoustics University of Stanford Stanford CA USA

This paper discusses a discrete-time modeling technique where the length of time delays can be arbitrarily adjusted. The new system is called a fractional delay waveguide model (FDWM). Formerly, FDWMs have only been implemented with FIR-type fractional delay filters. We show how an FDWM can be implemented using allpass filters. We use low-order allpass filters that are maximally-flat approximations of the ideal delay. The advantages of the allpass approach are computational efficiency and reduced approximation error. The proposed structure can be applied to discrete-time modeling of acoustic tubes, such as the human vocal tract or resonators of musical instruments.

关键词： Propagation delay Finite impulse response filter Acoustic waveguides Resonator filters Approximation error Digital filters Sampling methods Human voice Lattices Frequency response

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