• The Journal of the Acoustical Society of Korea
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• v.34 no.5
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• pp.394-402
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• 2015

CMA (Constant Modulus Algorithm) is one of the well-known algorithms in blind acoustic channel equalization. Generally, CMA converges slowly and the speed of convergence is dependent on a step-size in the CMA procedure. Many researches have tried to speed up the convergence speed by applying a variable step-size to CMA, e.g. the orthogonal projection CMA and algebraic optimal step-size CMA. In this paper, we summarize these two algorithms, and we propose a new CMA with improved convergence performance. The improvement comes from the switching between the orthogonal projection CMA and algebraic optimal step-size CMA. In simulation results, we show the performance improvement in the time invariant channels as well as in time varying channel.

• ETRI Journal
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• v.38 no.3
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• pp.435-443
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• 2016

In this paper, an efficient beam tracking algorithm for a regularized zero-forcing (RZF) approach in slowly fading multiple-input and single-output (MISO) broadcast channels is considered. By modifying an RZF equation, an RZF beam tracking algorithm is proposed using matrix perturbation theory. The proposed algorithm utilizes both beams from the previous time step and channel difference (between the previous and current time steps) to calculate the RZF beams. The tracking performance of the proposed algorithm is analyzed in terms of the mean square error (MSE) between a tracking approach and an exact recomputing approach, and in terms of the additional MSE caused by the beam tracking error at the receiver. Numerical results show that the proposed algorithm has almost the same performance as the exact recomputing approach in terms of the sum rate.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.27 no.6B
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• pp.569-578
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• 2002

This paper proposes a new multicarrier code division multiple access (CDMA) system. The proposed multicarrier CDMA system provides the advantages that the transmission bandwidth is more efficiently utilized by using divided spreading sequence, time and frequency diversity is achieved in frequency selective nultipath (acting channel, and inter-carrier interference (ICI) can be minimized by using specific data and code pattern. In this system, transmitted data bits are serial-to-parallel converted to some parallel branches. On each branch each bit is direct-sequence spread-spectrum modulated by divided spreading sequences and transmitted using orthogonal carriers. The receiver providers a Rake for each carrier, and the outputs of Rakes are combined to get time and frequency diversity. This multicarrier CDMA system allows additional flexibility in the choice of system parameters. Upon varying system parameters, bit error rate (BER) performance is examined for the proposed multicarrier CDMA system. Simulation results show that the proposed multicarrier CDMA scheme can achieve better performance than the other types of conventional multicarrier CDMA systems.

• The Journal of the Acoustical Society of Korea
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• v.18 no.1
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• pp.38-44
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• 1999

In this paper, a new spectral subtraction technique with two microphone inputs is proposed. In conventional spectral subtraction using a single microphone, the averaged noise spectrum is subtracted from the observed short-time input spectrum. This results in reduction of mean value of noise spectrum only, the component varying around the mean value remaining intact. In the method proposed in this paper, the short-time noise spectrum excluding the speech component is estimated by introducing the blocking matrix used in Griffiths-Jim-type adaptive beamformer with two microphone inputs, combined with the spectral compensation technique. A simulation was conducted to verify the effectiveness of the method.

• The Journal of the Acoustical Society of Korea
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• v.24 no.8
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• pp.454-461
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• 2005

Time spread due to the multipath in underwater acoustic channel, induces ISI (Inter-Symbol Interference) which degrades the performance of the underwater acoustic communication system. The passive phase conjugation (PPC) which improves the signal to multipath interference ratio (SMR) and therefore reduces the frequency selectivity. is a diversity communication technique giving a less ISI under multipath fading channel. Its frequency selectivity depends on the number of receiver array sensors and time varying source to receiver range. In this study, frequency selectivity of the PPC and its effects on bit error of underwater acoustic communication is analyzed by numerical simulation.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.25 no.6B
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• pp.1127-1135
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• 2000

The beamforming procedure in Adaptive Array Antenna System is affected by signal degradation and data rate due to DS-CDMA characteristics. Until this time, a lot of techniques are suggested to overcome this problems. This paper shows the simulation result about the beamforming performance of symbol level system that process slow data rate, compensated signal by despreading procedure in front of beamformer, and that of chip level system that process chip level signal without it. we analysis the performance using MSE, beam pattern, scattering points of beamformer output. chip level system is superior to symbol level system in time varying channel, while the performance of them didn't have difference in static channel.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.5 no.9
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• pp.1492-1512
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• 2011

Due to multiple hops, mobility and time-varying channel, supporting delay sensitive real-time traffic in wireless local area network-based (WLAN) mesh networks is a challenging task. In particular for real-time traffic subject to medium access control (MAC) layer control overhead, such as preamble, carrier sense waiting time and the random backoff period, the performance of real-time flows will be degraded greatly. In order to support real-time traffic, an efficient adaptive packet scheduling (APS) scheme is proposed, which aims to improve the system performance by guaranteeing inter-class, intra-class service differentiation and adaptively adjusting the packet length. APS classifies incoming packets by the IEEE 802.11e access class and then queued into a suitable buffer queue. APS employs strict priority service discipline for resource allocation among different service classes to achieve inter-class fairness. By estimating the received signal to interference plus noise ratio (SINR) per bit and current link condition, APS is able to calculate the optimized packet length with bi-dimensional markov MAC model to improve system performance. To achieve the fairness of intra-class, APS also takes maximum tolerable packet delay, transmission requests, and average allocation transmission into consideration to allocate transmission opportunity to the corresponding traffic. Detailed simulation results and comparison with IEEE 802.11e enhanced distributed channel access (EDCA) scheme show that the proposed APS scheme is able to effectively provide inter-class and intra-class differentiate services and improve QoS for real-time traffic in terms of throughput, end-to-end delay, packet loss rate and fairness.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.43 no.10 s.352
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• pp.69-78
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• 2006

High spectral efficiency can be obtained by adaptive modulation in which the modulation scheme is changed according to the channel environment. Thus it is especially suitable to mobile channel which is a typical example of time-varying channel. It is required to determine the optimum thresholds of signal-to-noise ratio(SNR) to change the modulation scheme effectively according to mobile speeds. Thus the optimum thresholds for specific mobile speeds to get the required bit error rate(BER) of $10^{-6}$ are obtained with the powerful turbo code in this paper. In addition, the optimum thresholds for the continuous mobile speed are proposed by interpolation of the obtained results. And the error performance and average spectral efficiency are investigated at various mobile speeds and channel environments.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.35 no.6C
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• pp.523-529
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• 2010

We propose a very low complexity lattice reduction (LR) algorithm for MIMO detection in time varying channels. The proposed scheme reduces the complexity by performing LR in a block-wise manner. The proposed scheme takes advantage of the temporal correlation of the channel matrices in a block and its impact on the unimodular matrices during LR process. From this, the proposed scheme can skip a number of redundant LR processes for consecutive channel matrices and performs a single LR in a block. The simulation results investigated in this letter reveal that the proposed detection scheme requires only 43.4% multiplications and 17.3% divisions of LLL-LR and only 50.2% multiplications and 68.2% divisions of the conventional adaptive LR with almost no performance degradation.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.4A
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• pp.421-430
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• 2006

In this paper, we propose a practical implementation of network-adaptive HD(high definition) MPEG-2 video streaming with a cross-layered channel monitoring(CLM) over the IEEE 802.11a WLAN(wireless local area network). For wireless channel monitoring, AP(access point) periodically measures the MAC(medium access control) layer transmission information and sends the monitoring information to a streaming server. This makes that the streaming server reacts more quickly as well as efficiently to the fluctuated wireless channel than that of the end-to-end monitoring(E2EM) scheme for the video adaptation. The streaming sewer dynamically performs the priority-based frame dropping to adjust the video sending rate according to the measured wireless channel condition. For this purpose, our streaming system nicely provides frame-based prioritized packetization by using a real-time stream parsing module. Various evaluation results over an IEEE 802.11a WLAM testbed are provided to verify the intended QoS adaptation capability The experimental results show that the proposed system can effectively mitigate the quality degradation of video streaming caused by the fluctuations of time-varying wireless channel condition.