• The Journal of the Acoustical Society of Korea
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• v.11 no.3
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• pp.53-60
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• 1992

Many adaptive beamforming methods have been studied for interference cancellation and speech signal enhancement in telephone conference and auditorium. Main aspect of adaptive beamforming methods for speech signal processing is different from radar, sonar and seismic signal processing because desire output signal should be apt to the human ear. Considering that phase of speech is quite insensible to the human ear, Sondhi proposed a nonlinear constrained optimization technique whose constraint was on the magnitude transfer function from the source to the output. In real environment the phase response of the speech signal affects the human auditorium system. So it is desirable to design linear phase system. In this paper, linear phase beamformer is proposed and sample processing algorithm is also proposed for real time consideration Simulation results show that the proposed algorithm yields more consistent beam patterns and deep nulls to the noise direction than Sondhi's.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.44 no.9
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• pp.1-8
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• 2007

In this paper, the new beamforming is proposed for an orthogonal frequency division multiplexing(OFDM) system with multi-input multi-output(MIMO). Through the proposed Pre-FFT beamforming technique for MIMO-OFDM, the multibeams are formed toward each multi-transmitter antenna of the desired user. The proposed beamforming for MIMO-OFDM can reduce cochannel interference and get diversity gain in the multi-user environment. Therefore, the performance of MIMO-OFDM system is very improved. BER performance improvement of the proposed approach is investigated through computer simulation by applying it to MIMO-OFDM system in the multi-user environment.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2000.10a
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• pp.468-472
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• 2000

This paper discusses the development of a acoustic vector channel simulator for the performance analysis of a acoustic digital communication system. The channel simulator consists of transmission module, acoustic channel model, receiver, beamformer, and adaptive equalizer. QPSK source signal is generated by the parameters specified by a user and transmitted. The transmitted signal generate multipath signals which have a different delay, amplitude, and dopper Sequency. The multipath signals with the acoustic noises is the received signal. We can analysis the communication system performance according to the antenna structure, beamforming algorithm, and equalization algorithm.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.14 no.5
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• pp.549-560
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• 1989

In this paper, the method of determining the initial weighting vlaues fuctioning as a filter under the Directional Constrained Minimization of Power(DCMP) algorithm is presented. By analyzing the sideband beamformer with the Finite Impulse Response (FIR) filter concepts, the constraints of any desired directions are obtained and the initial weighing values with fast adaptation time are formulated from those constraints. By applying this proposed initial weighting values to the DCMP and the spatial averaging processor, the interference of a desired direction and the coherent noises are eliminated at the same time. The improvement of this method compared with the existing algorithm is confirmed by computer simulation.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.37 no.1
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• pp.83-89
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• 2000

Antenna arrays at base-stations can be used to transmit and receive information selectively in space by reducing the interference effects. In this paper, a new model of locally scattered signals in the vicinity of mobiles is proposed, and under this model the weights of the beamformer are obtained. Computer simulation results demonstrate that the proposed scheme shows an excellent performance and works well even in the urban environment where there exist many multipath propagations with wide angular spread.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2003.10a
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• pp.296-299
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• 2003

In this paper, an underwater acoustic digital transceiver is designed and implemented by a multiple DSPs system. We have designed a QPSK transmitter based on look-up table and 13-symbols Barker code is used for frame synchronization. Channel distortions are compensated by a wide-band beamformer based on FIR filter and an adaptive equalize. with RLS algorithm. Uniform linear array (ULA) with four elements is used for the spartial signal processing. 1/2 convolutional code and Viterbi decoder are implemented to overcome time-varying multi-path fading. Also, we show experimental results in the underwater anechoic basin at KRISO/KORDl and Goseong, Donghae and Soyang lake of Kangwon-do.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.4 no.4
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• pp.861-868
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• 2000

This paper discusses the development of an acoustic vector channel simulator for the performance analysis of an acoustic digital communication system. The channel simulator consists of transmission module, acoustic channel model, receiver, beamformer, and adaptive equalizer. QPSK source signal is generated by the parameters specified by a user. The transmitted signal generates multipath signals which have a different delay, amplitude, and dopper frequency. The multipath singnals with the acoustic noises are the received signal. This paper presents the performance analysis of an acoustic digital communication system according to the antenna structure and the various baseband signal processing techniques.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.17 no.1
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• pp.32-40
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• 2013

In this paper, we propose a multi-channel speech enhancement algorithm using beamforming and direction-of-arrival (DOA)-based masking. The proposed algorithm enhances noisy speech basically by the linearly constrained minimum variance (LCMV) algorithm and then a mel-scale Wiener filter designed using DOA-based masking is applied to remove still remaining noises. To improve the performance, we optimize the learning rate of the adaptive filters in LCMV and the DOA threshold to detect target speech spectrum. As performance indices, the perceptual evaluation of speech quality (PESQ) score and output SNRs are measured. Experimantal results show that the proposed algorithm outperforms the conventional LCMV beamformer by 0.09 in PESQ score and 5.75 dB in output SNR, respectively.

• The Journal of the Korea institute of electronic communication sciences
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• v.9 no.2
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• pp.195-202
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• 2014

The Global Positioning System (GPS), which has various military and commercial applications, is designed to estimate the location of the specific user or object. In order to accurately estimate the location, GPS requires at least four satellite signals. The GPS receiver operates on extremely low signal-to-noise ratio (SNR) environment and it may suffer from various interference signals with the extremely high power. In this paper, we introduce a blind adaptive receiver based on the modified despreader, which suppress interference signals and detect GPS signals of interest without requiring explicit angle-of-arrival (AOA) information. We, also, provide the mathematical analysis for the signal-to-interference and noise ratio (SINR) of the modified despeader beamformer output. A representative computer simulation example is presented to illustrate the interference suppression performance of the considered GPS receiver and mathematical analysis of the SINR.

• The Journal of the Acoustical Society of Korea
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• v.23 no.2
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• pp.117-124
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• 2004

In this paper, we have been implemented the QPSK-based underwater transmission systems using DSP in order to transmit the underwater image data. We have adopted a BDPA (Block Data Parallel Architecture) to control multiple DSPs used in the transmitter and receiver in order to transmit the image data in real-time. We also have developed GUI software in order to drive and to debug the implemanted system in real-time. We have executed open sea tests in order to analyze the performance of the implemented system at East Sea near Kosung in Kangwon-Do. As a result of these experiments, it has been demonstrated that 10 kbps image data can be received without errors at 30m and 80m depth points, while the distance between the transmitter and the receiver is up to 20m.