• The Journal of Korean Institute of Communications and Information Sciences
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• v.39C no.12
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• pp.1346-1351
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• 2014

We propose a novel approach for stereophonic acoustic echo suppression using spectro-temporal filtering based on soft decision. Unlike the conventional approaches estimating the echo pathes directly, the proposed technique can estimate stereo echo spectra without any double-talk detector. In order to improve the estimation of echo spectra, the extended power spectrum density matrix and echo overestimation control matrix are applied on this method. In addition, this echo suppression technique is based on soft decision technique using speech absence probability in STFT domain. Experimental results show that the proposed method improves compared with the conventional approaches.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.19 no.4
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• pp.25-33
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• 2018

In this paper, we propose an iterative self-interference (SI) channel estimation method for in-band full-duplex cellular systems that employ orthogonal frequency division multiple access (OFDMA) on downlink (DL) and single-carrier frequency division multiple access (SC-FDMA) on uplink (UL), as in Long Term Evolution (LTE) systems. The proposed method first acquires coarse estimates of SI channels using DL signals and UL pilots, which are known to the base stations, and then refines the estimates by consecutively exploiting averaging in the frequency domain and channel truncation in the time domain. In addition, the method enhances the estimates further by iteratively executing this estimation procedure, and does not require any radio resources dedicated to SI channel estimation. Simulation results demonstrate that by significantly improving the SI channel estimation performance without requiring exact knowledge of the SI channel length, the proposed method achieves UL channel estimation performance and signal-to-interference-plus-noise ratio (SINR) performance very close to those in perfect SI cancellation.

• Journal of Convergence for Information Technology
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• v.10 no.10
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• pp.24-31
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• 2020

In non-orthogonal multiple access (NOMA), the degraded performance of the weaker channel gain user is a problem. In this paper, we propose the asymmetric binary pulse amplitude modulation (2PAM), to improve the bit-error rate (BER) performance of the weaker channel user in NOMA with the tolerable BER loss of the stronger channel user. First, we design the asymmetric 2PAM, calculate the total allocated power, and derive the closed-form expression for the BER of the proposed scheme. Then it is shown that the BER of the weaker channel user improves, with the small BER loss of the stronger channel user. The superiority of the proposed scheme is also validated by demonstating that the signal-to-noise ratio (SNR) gain of the weaker channel user is about 10 dB, with the SNR loss of 3 dB of the stronger channel user. In result, the asymmetric 2PAM could be considered in NOMA of 5G systems. As a direction of the future research, it would be meaningful to analyze the achievable data rate for the propsed scheme.

• The Journal of the Acoustical Society of Korea
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• v.11 no.2
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• pp.73-84
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• 1992

This paper studies noise subtraction methods and distance measures for speech recognition in a noisy environment, and investigates noise robustness of the distance measures applied to the problem of isolated word recognition in white Gaussian and colored noise (vehicle noise) environments. Noise subtraction methods which can be used as a pre-processor for the speech recognition system, such as the spectral subtraction method, autocorrelation subtraction method, adaptive noise cancellation and acoustic beamforming are studied, and distance measures such and Log Likelihood Ratio ($d_{LLR}$), cepstral distance measure ($d_{CEP}$), weighted cepstral distance measure ($d_{WCEP}$), spectral slope distance measure ($d_{RPS}$) and cepstral projection distance measure ($d_{CP},\;d_{BCP},\;d_{WCP},\;d_{BWCP}$) are also investigated. Testing of the distance measures for speaker-dependent isolated word recognition in a noisy environment indicate that $d_{RPS}\;and\;d_{WCEP}$ which weigh higher order cepstral coefficients more heavily give considerable performance improvement over $d_{CEP}and\;d_{LLR}$. In addition, when no pre-emphasis is performed, the recognizer can maintain higher performance under high noise conditions.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.38 no.12
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• pp.10-19
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• 2001

In this paper, we propose an effective but simple multi-rate parallel interference canceller(PIC) for the international mobile telecommunications-2000(IMT-2000) 3rd generation partnership project (3GPP) system. For effective multi-rate processing, we define the basic block as one symbol period of the dedicated physical control channel(DPCCH) having the lowest data rate and common to all users. Then, decision and interference cancellation are performed at every basic block. For an asynchronous channel, we propose an advance removal scheme that removes in advance multiple access interference(MAI) due to the next blockof other users with shorter delay. Introducing a pipeline structure at a sample base, we can implement efficiently the PIC using the advance removal scheme with a minimum hardware and no extra computations. Through computer simulations, we analyze the bit error rate(BER) performance of the proposed PIC with respect to signal-to-noise ratio(SNR) and the number of users.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.28 no.4
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• pp.298-308
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• 2017

This paper presents a Interleaved Buck Converter(IBC) system with Random PWM to reduce electromagnetic noise by harmonics. Swithced mode power supply generally controlled by high switching frequency have a electromagnetic interference(EMI) issue due to the high-voltage/high-current switching to regulate the voltage in buck converter. To solve the problem. we present a novel IBC system with PRBS. IBC system has two active switches with 180 phase difference that controll the cicuit with two PWM signal. IBC system may be disadventageous for the cost due to the addtion of one set of switch, but it has adventages of power distribution, current ripple cancellation, fast transient response, and passive component size reduction. To verify the validity of study, simulation program has been bulit using PSIM and the experimental results of IBC system using RPWM was compared with the conventinal PWM and randomized PWM.

• The Journal of the Acoustical Society of Korea
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• v.20 no.6
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• pp.17-25
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• 2001

Traditional adaptive noise cancelling methods rely their performance on various interfering parameters, such as convergence speed, tracking ability, numerical stability, relative frequency characteristics between target and reverberation signals, and activity of the target. In this paper, an adaptive noise cancelling method is suggested, which Provides a successful tradeoff mon these factors. It is designed to work on the transform domain, adopts the Gram-Schmidt orthogonalization process, and is implemented by the escalator algorithm. The transform domain approach supports a tradeoff between the convergence speed and numerical cost. The proposed method is verified by applying a real-data collected in the shallow waters off the east coasts of korea. It is shown that it has a good reverberation-rejection capability even for the target signal with adjacent frequency components to those of the reverberation, and its performance is invariant for the activity of the target.

• Journal of Communications and Networks
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• v.13 no.3
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• pp.232-239
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• 2011

This paper investigates the user selection problem of successive zero-forcing precoded multiuser multiple-input multiple-output (MU-MIMO) downlink systems, in which the base station and mobile receivers are equipped with multiple antennas. Assuming full knowledge of the channel state information at the transmitter, dirty paper coding (DPC) is an optimal precoding strategy, but practical implementation is difficult because of its excessive complexity. As a suboptimal DPC solution, successive zero-forcing DPC (SZF-DPC) was recently proposed; it employs partial interference cancellation at the transmitter with dirty paper encoding. Because of a dimensionality constraint, the base station may select a subset of users to serve in order to maximize the total throughput. The exhaustive search algorithm is optimal; however, its computational complexity is prohibitive. In this paper, we develop two low-complexity user scheduling algorithms to maximize the sum rate capacity of MU-MIMO systems with SZF-DPC. Both algorithms add one user at a time. The first algorithm selects the user with the maximum product of the maximum column norm and maximum eigenvalue. The second algorithm selects the user with the maximum product of the minimum column norm and minimum eigenvalue. Simulation results demonstrate that the second algorithm achieves a performance similar to that of a previously proposed capacity-based selection algorithm at a high signal-to-noise (SNR), and the first algorithm achieves performance very similar to that of a capacity-based algorithm at a low SNR, but both do so with much lower complexity.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.13 no.12
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• pp.5862-5887
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• 2019

In this paper, we propose a two-way relaying scheme using non-orthogonal multiple access (NOMA) technology. In this scheme, two sources transmit packets with each other under the assistance of the decode-and-forward (DF) relays, called as a TWDFNOMA protocol. The cooperative relays exploit successive interference cancellation (SIC) technique to decode sequentially the data packets from received summation signals, and then use the digital network coding (DNC) technique to encrypt received data from two sources. A max-min criterion of end-to-end signal-to-interference-plus-noise ratios (SINRs) is used to select a best relay in the proposed TWDFNOMA protocol. Outage probabilities are analyzed to achieve exact closed-form expressions and then, the system performance of the proposed TWDFNOMA protocol is evaluated by these probabilities. Simulation and analysis results discover that the system performance of the proposed TWDFNOMA protocol is improved when compared with a conventional three-timeslot two-way relaying scheme using DNC (denoted as a TWDNC protocol), a four-timeslot two-way relaying scheme without using DNC (denoted as a TWNDNC protocol) and a two-timeslot two-way relaying scheme with amplify-and-forward operations (denoted as a TWANC protocol). Particularly, the proposed TWDFNOMA protocol achieves best performances at two optimal locations of the best relay whereas the midpoint one is the optimal location of the TWDNC and TWNDNC protocols. Finally, the probability analyses are justified by executing Monte Carlo simulations.

• The Journal of The Korea Institute of Intelligent Transport Systems
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• v.10 no.4
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• pp.71-77
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• 2011

In this paper, we propose a conversational quality measurement (CQM) system for providing the objective QoS of high quality mobile VoIP voice telecommunication. For measuring the conversational quality, the VoIP telecommunication system is implemented in two smart phones connected with VoIP. The VoIP telecommunication system consists of echo cancellation, noise reduction, speech encoding/decoding, packet generation with RTP (Real-Time Protocol), jitter buffer control and POS (Play-out Schedule) with LC (loss Concealment). The CQM system is connected to a microphone and a speaker of each smart phone. The voice signal of each speaker is recorded and used to measure CE (Conversational Efficiency), CS (Conversational Symmetry), PESQ (Perceptual Evaluation of Speech Quality) and CE-CS-PESQ correlation. We prove the CQM system by measuring CE, CS and PESQ under various SNR, delay and loss due to IP network environment.