• The Journal of the Korea institute of electronic communication sciences
• /
• v.17 no.5
• /
• pp.995-1002
• /
• 2022

Global Navigation Satellite System(GNSS) is a convenient system that acquires position and time information of a receiver if only satellite signals can be received anywhere in the world. However navigation signals include errors and a position error occurs according to the reception state of the signal. Also, a position error is affected by the geometric arrangement of the satellites. Therefore a receiver position performance varies by the number and status of visible satellites The condition of satellite signals is not good when the satellite rises or sets and the position change of receiver occurs when the signal is blocked by an obstacle such as a building in the urban area. In this paper, we proposed methods to improve the GNSS performance by using pseudorange correction method estimating the correction amount and the visible satellites selection method. By applying the proposed methods to an environment in which the number of visible satellites changes variously, the performance enhancement was verified.

• The Transactions of the Korean Institute of Electrical Engineers A
• /
• v.54 no.6
• /
• pp.293-305
• /
• 2005

This paper describes an improved Prony method in its speed, accuracy and reliability by efficiently determining the optimal sampling interval with use of DFT (discrete Fourier transformation). In the Prony method the computation time is dominated by the size of the linear prediction matrix, which is given by the number of data times the modeling order The size of the matrix in a general Prony method becomes large because of large number of data and so does the computation time. It is found that the Prony method produces satisfactory results when SNR is greater than three. The maximum sampling interval resulting minimum computation time is determined using the fact that the spectrum in DFT is inversely proportional to sampling interval. Also the process of computing the modes is made efficient by applying Hessenberg method to the companion matrix with complex shift and computing selectively only the dominant modes of interest. The proposed method is tested against the 2003 KEPCO system and found to be efficient and reliable. The proposed method may play a key role in monitoring in real time low frequency oscillations of power systems .

• KSII Transactions on Internet and Information Systems (TIIS)
• /
• v.7 no.10
• /
• pp.2338-2356
• /
• 2013

In cognitive radio networks, acquiring the position information of the primary user is critical to the communication of the secondary user. Localization of primary users can help improve the efficiency with which the spectrum is reused, because the information can be used to avoid harmful interference to the network while simultaneity is exploited to improve the spectrum utilization. Despite its inherent inaccuracy, received signal strength based on range has been used as the standard tool for distance measurements in the location detection process. Most previous works have employed the path-loss propagation model with a fixed value of the path loss exponent. However, in actual environments, the path loss exponent for each channel is different. Moreover, due to the complexity of the radio channel, when the number of channel increases, a larger number of RSS measurements are needed, and this results in additional energy consumption. In this paper, to overcome this problem, we propose using the Bayesian compressive sensing method with a calibrated path loss exponent to improve the performance of the PU localization method.

• Phonetics and Speech Sciences
• /
• v.1 no.1
• /
• pp.63-67
• /
• 2009

In this paper, in order to protect music copyrights, we propose a music identification system which is scalable to the number of pieces of registered music and robust to signal-level variations of registered music. For its implementation, we define the new concepts of 'music word' and 'music phoneme' as recognition units to construct 'music acoustic models'. Then, with these concepts, we apply the HMM-based framework used in continuous speech recognition to identify the music. Each music file is transformed to a sequence of 39-dimensional vectors. This sequence of vectors is represented as ordered states with Gaussian mixtures. These ordered states are trained using Baum-Welch re-estimation method. Music files with a suspicious copyright are also transformed to a sequence of vectors. Then, the most probable music file is identified using Viterbi algorithm through the music identification network. We implemented a music identification system for 1,000 MP3 music files and tested this system with variations in terms of MP3 bit rate and music speed rate. Our proposed music identification system demonstrates robust performance to signal variations. In addition, scalability of this system is independent of the number of registered music files, since our system is based on HMM method.

• The Journal of the Acoustical Society of Korea
• /
• v.35 no.5
• /
• pp.395-402
• /
• 2016

In this paper, a wideband adaptive beamforming approach for an acoustic vector sensor linear array is presented. It is a very important issue to estimate the stable covariance matrix for adaptive beamforming. In the conventional wideband adaptive beamforming based on coherent signal-subspace (CSS) processing, the error of bearing estimates is resulted from the focusing matrix estimation and the large number of data snapshot is necessary. To alleviate the estimation error and snapshot deficiency in estimating covariance matrix, the steered covariance matrix method in the pressure sensor is extended to the vector sensor array, and the subarray technique is incorporated. By this technique, more accurate azimuth estimates and a stable covariance matrix can be obtained with a small number of data snapshot. Through simulation, the azimuth estimation performance of the proposed beamforming method and a wideband adaptive beamforming based on CSS processing are assessed.

• Journal of Advanced Navigation Technology
• /
• v.16 no.2
• /
• pp.297-306
• /
• 2012

In this paper, we propose a Time Delay Traceback Scheme for the TDOA location estimation performance enhancement in NLOS environment and analyze the performance in various conditions. We place multiple readers in a square($300m{\times}300m$) searching area for reuse of received signal. Also, we use more active NLOS reader detection methode for NLOS error mitigation. when NLOS time delay 70 m, the number of the NLOS reader is 3 and the received sub-blinks number 3, proposed time delay trace-back scheme improve the RMSE about 16 m. From these results, we confirm that the proposed time delay traceback scheme is well-suited for the high precision location estimation to offer the location based service.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.32 no.1C
• /
• pp.23-33
• /
• 2007

Video compression is essential technique for fast and efficient transmission of a large amount of video data maintaining high quality. Also, motion estimation and motion compensation is most important technique for efficient video compression. A proposed method is improved diamond search method which uses split diamond pattern and rotated diamond pattern. In particular, the proposed method shows superb result when it is used for the sequence with a direction of camera moving. Moreover when it is used for the sequence with little motion, complexity is reduced considerably by using fewer search points. Also, by varying the number of initial search points, the propose method can provide several options in terms of duality or speed. Simulation results shows that the proposed method sustains better visual quality compared with diamond search method and HEXBS even by using fewer search points. Besides, compared with existing methods, it is able to conduct a motion estimation more efficiently by changing the number of search points adaptively according to motion of video data.

• The Journal of the Korea institute of electronic communication sciences
• /
• v.13 no.6
• /
• pp.1207-1212
• /
• 2018

The signal intelligence (SIGINT) technology is actively used for collecting various data, in a number of fields, including a military industry. In order to collect the signal information and data and to transmit/receive the collected data efficiently, the accurate angle-of-arrival (AOA) information is required and communication disturbance from the interference or jamming signal should be minimized. In this paper, we present the structure of an adaptive beam-forming satellite system based on the planar array antenna, for collecting and transmitting/receiving the signal information and data efficiently. The presented adaptive beam-forming system consists of an antenna in the form of a planar array, an AOA estimator based on the Multiple Signal Classification (MUSIC) algorithm, an adaptive Minimum Variance Distortionless Response (MVDR) interference canceler, a signal processing and D/B unit, and a transmission beamformer based on Minimum mean Square Error (MMSE). In addition, through the computer simulation, we evaluate and analyze the performance of the proposed system.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.29 no.7A
• /
• pp.794-806
• /
• 2004

This paper deals with correction of frequency offset and analysis of quantization angle noise in the IEEE 802.1la OFDM system. The rotation phase per symbol due to the carrier frequency offset is estimated from auto-correlation of the short Preambles, which are over-sampled for the reduction of noise in OFDM signals. The pilot signals are introduced to estimate the rotation phase per OFDM symbol due to estimation error of the carrier frequency offset and the sampling frequency onset. During the estimation and correction of the frequency onsets, a CORDIC processor and a look-up table are used for the conversion between a rotation phase and its complex number. Being calculated by a limited number of bits in the CORDIC processor and the look-up table, the rotation phase and its complex number have quantization angle errors. The quantization errors are analyzed as SNR (signal to noise ratio) due to the quantization bit numbers. The minimum bit number is suggested to meet the specification of IEEE 802.1la properly. Finally, the quantization errors are evaluated through simulations on number of quantization bits and SNR of received signals.

• Journal of the Institute of Electronics Engineers of Korea TC
• /
• v.48 no.4
• /
• pp.15-23
• /
• 2011

An interleaver rearranges a channel-encoded data in the symbol unit to spread burst errors occurred in channels into random errors. Thus, the interleaving process makes it difficult for a receiver, who does not have information of the interleaver parameters used in the transmitter, to de-interleave an unknown interleaved signal. Recently, various researches on the reconstruction of an unknown interleaved signal have been studied in many places of literature by estimating the interleaver parameters. They, however, have been mainly focused on the estimation of the block interleaver parameters required to reconstruct the de-interleaver. In this paper, as an extension of the previous researches, we estimate the convolutional interleaver parameters, e.g., the number of shift registers, a shift register depth, and a codeword length, required to de-interleave the unknown data stream, and propose the de-interleaving procedure by reconstructing the de-interleaver.