• Journal of rehabilitation welfare engineering & assistive technology
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• v.8 no.1
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• pp.41-46
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• 2014

In this paper, we propose a novel voice activity detection (VAD) algorithm using feature vectors based on TE (teager energy). Specifically, power spectral deviation (PSD), which is used as the feature for the VAD in the IS-127 noise suppression algorithm, is obtained after the input signal is transfomed by Teager energy operator. In addition, the TE-based likelihhod ratio are derived in each frame to modifiy the PSD for further VAD. The performance of our proposed VAD algorithm are evaluated by objective testing (total error rate, receiver operating characteristics, perceptual evaluation of speech quality) under various environments, and it is found that the proposed method yields better results than conventional VAD algorithms in the non-stationary noise environments under 5 dB SNR (total error rate = 2.6% decrease, PESQ score = 0.053 improvement).

• International Journal of Railway
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• v.3 no.2
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• pp.39-45
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• 2010

The RF-CBTC (Radio Frequency-Communication Based Train Control) System is a communication system in railroad systems. The communication method of RF-CBTC system is the wireless between the wayside device and on-board device. The wayside device collects its location and speed from each train and transmits the distance from the forwarding train to the speed-limit position to it. The on-board device controlling device controls the speed optimum for the train. In the case of the RF-CBTC system used in Korea, transmission frequency is 2.4 [GHz]. It is the range of ISM(Industrial Scientific and Medical equipment) band and transmission of voice and data is performed by CDMA (Code Division Multiple Access) method. So noises are made in the AWGN (Additive White Gaussian Noise) and fading environment. Currently, the SNR (Signal to Noise Ratio) is about 20 [dB], so due to bit errors made by noises, transmission of reliable information to the train is not easy. Also, in the case that two tracks are put to a single direction, it is needed that two trains transmit reliable voice and data to a wayside device. But, by noises, it is not easy that just a train transmits reliable information. In this paper, we estimated the BER (Bit Error Rate) related to the SNR of voice and data transmission in the environment such as AWGN and fading from the RF-CBTC system using the CDMA method. Also, we supposed the SNR which is required to meet the BER standard for voice and data transmission. By increasing the processing gain that is a ratio of chip transmission to voice and data transmission, we made possible voice and data transmission from maximally two trains to a wayside device, and demonstrated it by using Matlab program.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.10 no.1
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• pp.104-110
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• 2006

The advantage of the DSM which employ multi-bit quantizer is the increased SNR at the modulator's output. Typically 6 dB improvement is effected for every one additional bit. But multi-bit quantizer evidently requires multi-bit DAC in the feedback loop. The integral linearity error of the feedback DAC has direct impact upon the system performance and degraded SNR of the system. In order to mitigate the negative impact the DAC has on the system performance, many DEM(Dynamic Element Matching) schemes has been proposed. Among the proposed schemes, four schemes(DER,CLA,ILA,DWA) are explained and its performance has been compared. DWA(Data Weighted Averaging) method shows the best performance of the all.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.44 no.2 s.314
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• pp.102-107
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• 2007

Since the Matrix Pencil (MP) method can be performed to estimate multiple DOAs by using only single snapshot, this method is suitable for short data length or when the environment is dynamic. As the number of array increases, the computational load increases due to complex number computation. This paper presents an approach based on a unitary matrix pencil (MP) algorithm to reduce the computational load. Unitary transformation for the MP method has been suggested and formulated successfully. The computer simulation shows that the error rate of proposed method agree with that of MP for different SNR values.

• Journal of Navigation and Port Research
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• v.29 no.3 s.99
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• pp.245-250
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• 2005

In this paper, considering PAR of transmitter which is pointed out OFDM system's problem, we designed Coded OFDM (COFDM) system and estimated BER and SNR using PAR reduction method In order to evaluate performance, we compared M-ary PSK (M-ary Phase Shift Keying) with M-ary QAM (M-ary Quadrature Amplitude Modulation). In result, performance of 16-PSK and 16-QAM came to good Moreover, 16-QAM showed better performance of about 2 dB in 10-3 error probability and performance of about 5 dB in Peak power clipping than that of 16-PSK.

• Journal of Satellite, Information and Communications
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• v.10 no.2
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• pp.19-23
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• 2015

In this paper, we present the return link ACM method to improve the link availability and system throughput for satellite communication service. Also, we describe the optimization of an algorithm for channel prediction using the LMS (Least Mean Square) adaptive filter and the MODCOD (Modulation & Code rate) decision. The simulation results show that the optimized filter taps and step-size of adaptive filter are 2 and 0.00026, respectively. And also confirms the required SNR margin for minimization of MODCOD decision error is 0.3dB.

• Journal of Communications and Networks
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• v.7 no.3
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• pp.243-247
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• 2005

We propose an efficient block least-mean-square (BLMS) adaptive algorithm, in conjunction with error control coding, for direct-sequence code division multiple access (DS-CDMA) systems. The proposed adaptive receiver incorporates decision feedback detection and channel encoding in order to improve the performance of the standard LMS algorithm in convolutionally coded systems. The BLMS algorithm involves two modes of operation: (i) The training mode where an uncoded training sequence is used for initial filter tap-weights adaptation, and (ii) the decision-directed where the filter weights are adapted, using the BLMS algorithm, after decoding/encoding operation. It is shown that the proposed adaptive receiver structure is able to compensate for the signal-to­noise ratio (SNR) loss incurred due to the switching from uncoded training mode to coded decision-directed mode. Our results show that by using the proposed adaptive receiver (with decision feed­back block adaptation) one can achieve a much better performance than both the coded LMS with no decision feedback employed. The convergence behavior of the proposed BLMS receiver is simulated and compared to the standard LMS with and without channel coding. We also examine the steady-state bit-error rate (BER) performance of the proposed adaptive BLMS and standard LMS, both with convolutional coding, where we show that the former is more superior than the latter especially at large SNRs ($SNR\;\geq\;9\;dB$).

• The Journal of Korean Institute of Communications and Information Sciences
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• v.21 no.12
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• pp.3174-3182
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• 1996

In this paper, a new receiver is proposed for the detection of the OFDM(orthogonal frequency division multiplexing) signals in the time-selective multipath fading channel. For the optimal detection, we estimate the transmitted symbols from OFDM demultiplexing signal using the LMLE(linear masimum likelihold estimation) algorithm. Also, in this paper, the lowerbound for BER(bit error rate) using Taylor series approximation is provided. If the matched filter is used for the OFDM receiver in the time-selectivemultipath fading channel, it is known that the SER(symbol error rate) is always greater than $10^{-1}$, due to the cross-talk between adjacent channels. But, the proposed receiver provides of SER with 15dB SNR. Also, it is found that for the receiver implemented using the LMLE algorithm, the performance is shown to be not affected by the increase of th enumber of subchannel and channel path.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.35 no.5C
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• pp.432-438
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• 2010

In this paper we study the effect of bit-to-symbol mappings on the convergence behavior of turbo equalizers employing low-density parity-check (LDPC) codes over high order modulations. We analyze the effective SNR of the outputs from linear minimum mean-squared error (MMSE) equalizers and the convergence property of LDPC decoding for different symbol mappings. Numerical results show that the bit-reliability (BR) mapping provides better performance than random mapping in LDPC-coded turbo equalizers over high order modulations. We also verify the effect of symbol mappings through the noise threshold and error performance.

• The Journal of the Acoustical Society of Korea
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• v.13 no.5
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• pp.77-82
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• 1994

In speech signal processing, it is very important to detect the pitch exactly. Although various methods for detecting the pitch of speech signals have been developed, it is difficult to exactly extract the pitch for wide range of speakers and various utterances. Thus we propose a new pitch detection algorithm which takes advantage of the G-peak extraction. It is a method to detect the pitch period of the voiced signals by finding MZCI (maximum zero-crossing interval) of the G-peak which is defined as cut-off bandwidth rate of LPF (low pass filter). This algorithm performs robustly with a gross error rate of 3.63% even in 0 dB SNR environement. The gross error rate for clean speech is only 0.18%. Also it is able to process all courses with high speed.