• Journal of the Korean Institute of Telematics and Electronics
• /
• v.18 no.1
• /
• pp.41-50
• /
• 1981

The human speech contains many redundancies. To economize communication channel or memory size for a computerized synthesis of human voices, it is necessary to compress the data before sending. We have treated human speech organ as an eighth order dynamic system which is time varying as the person speaks. Using an anaylyzer of our design, each eight parameters are obtained for the vowels [아], [어], [오], [우], [으], [이], [애], and (외) of korean language with considerable discrepancies between persons. Supplying those parameters to a synthesizer which we have made, we have sucoeeded in the simulation of human speech for the above mentioned vowels of Korean language and observed that they bear all the features of the original speakers.

• Journal of The Institute of Information and Telecommunication Facilities Engineering
• /
• v.8 no.4
• /
• pp.201-206
• /
• 2009

The third generation mobile communications system requiring the reliable multimedia data transmission has provided with the reliable voice, data and video services over the variable propagation environment. However the broadband wireless multiple access technologies cause Inter Symbol Interference (ISI) or Multiple Access Interference (MAI) to degrade the performance of W-CDMA (Code Division Multiple Access) system. Constant Modulus Algorithm (CMA) which is frequently used as the adaptive blind equalizers to remove the interfering signal has ill-convergence phenomenon without proper initialization. In this paper, new blind equalization method based on conventional CMA is proposed to improve the channel efficiency, and through computer simulation this is tested over the time varying fading environment of mobile communication system. Consequently, new blind equalization method into concatenated Kalman filter with CMA is verified better than conventional CMA through adopting minimum mean square errors and eye-pattern obtained from algorithm are compared.

• KSII Transactions on Internet and Information Systems (TIIS)
• /
• v.9 no.5
• /
• pp.1587-1605
• /
• 2015

Mobile cloud computing (MCC) combines mobile Internet and cloud computing to improve the performance of applications. In MCC, the data processing and storage for mobile devices (MDs) is provided on the remote cloud. However, MCC faces the problem of energy efficiency caused by randomly varying channels. In this paper, by introducing the Lyapunov optimization method, we propose a combined service subscription and delivery (CSSD) algorithm that can guide the users to subscribe to services reasonably. This algorithm can also determine whether to deliver the data and to whom data is sent in the current time unit based on the queue backlog and the channel state. Numerical results validate the correctness and effectiveness of our proposed CSSD algorithm.

• International Journal of Internet, Broadcasting and Communication
• /
• v.15 no.1
• /
• pp.1-7
• /
• 2023

TCP is considered a major candidate transport protocol even for constrained IoT networks due to its ability to integrate into the existing network infrastructures. Since TCP implementations such as uIP TCP often allow only a single TCP segment per connection to be unacknowledged at any given time due to resource constraints, the congestion control relies only on RTO management. In our previous work, to address the problem that uIP TCP performs poorly particularly when a radio duty cycling mechanism is enabled and the hidden terminal problem is severe, we proposed a RTO scheme for uIP TCP and validated the performance through Cooja simulation. In this study, we investigate the effect of other factors that were not considered in our previous work. More specifically, the effect of traffic intensity, the degree of the hidden terminal problem, and RDC is investigated by varying the offered load and the transmission range, and the RDC channel check rate. Simulation results imply that we need to further investigate how to improve TCP performance when the radio duty cycling mechanism is used.

• The Journal of the Acoustical Society of Korea
• /
• v.35 no.4
• /
• pp.295-302
• /
• 2016

Underwater acoustic communication channel is often defined as a multipath fading channel since the multipath arrivals from various paths interfere with each other and cause frequency dependent constructive or destructive interference in received signals. Therefore signal-to-noise ratio (SNR) of received signal fluctuates as a function of frequency. In addition, sea surface fluctuation induces frequency dependent time variant signal fading due to coherent component variation of surface bounce path. The frequency shift keying (FSK) system is known to be less sensitive and more robust under these interference and fading, and M-ary frequency shift keying (MFSK) system is adopted to increase a data rate. In this study, a bit error rate (BER) of 4 channels 4FSK system are examined in shallow sea multipath channel. Experimental results show that RS code reduces efficiently the BER of 4FSK system since frequency dependent time-varying fading is characterized to give burst errors. The BER of a different data rate or different source-to-receiver range depends on not only the channel coherent bandwidth but also frequency dependent multipath fading.

• The Journal of the Acoustical Society of Korea
• /
• v.39 no.1
• /
• pp.8-15
• /
• 2020

In this paper, we propose a frame synchronization algorithm for robust to the combined effects of large Doppler fluctuations and extended, time-varying multipath in the underwater acoustic communication. From the algorithm, we can recover a high timing error which is occurred from an acoustic propagation delay and uncertainty of oscillator between transmitter and receiver. In order to verify the performance of the synchronization algorithm, the lake trial results are used. The lake experiments are performed in a Gyeongcheonho located in Mungyeong-si, Gyeongsangbuk-do. We can see that the start position of frame is adjusted after the frame synchronization while the receiver moving.

• The Journal of the Acoustical Society of Korea
• /
• v.31 no.1
• /
• pp.1-10
• /
• 2012

The linear and decision-feedback equalization can mitigate time-varying intersymbol interference (ISI) caused by time-varying multipath propagation for underwater acoustic channels. The perfect elimination of interference components, however, is difficult using the linear equalization and the decision feedback equalizer has an error propagation problem. To overcome these shortcomings, this paper proposes an equalizer mode selection method using training sequences. The proposed method selects an equalization mode corresponding to the signal-to-noise ratio (SNR). If the SNR is low, the proposed system operates the linear equalizer for preventing the error propagation and if the SNR is high, the decision feedback equalizer for eliminating the residual ISI. Therefore, the proposed method can improve the error performance compared to the conventional equalizers. The computer simulation shows the proposed method improves the bit error performance using practical underwater channels responses acquired from the sea experiment.

• 전자공학회논문지 IE
• /
• v.43 no.2
• /
• pp.80-86
• /
• 2006

The wireless communication based on LDPC and adaptive spatial-subcarrier coded modulation using MQAM for orthogonal frequency division multiplexing (OFDM) wireless transmission by using instantaneous channel state information and employing multiple antennas at both the transmitter and the receiver. Adaptive coded modulation is a promising idea for bandwidth-efficient transmission on time-varying, narrowband wireless channels. On power limited Additive White Gaussian Noise (AWGN) channels, low density parity check (LDPC) codes are a class of error control codes which have demonstrated impressive error correcting qualities, under some conditions performing even better than turbo codes. The paper demonstrates OFDM with LDPC and adaptive modulation applied to Multiple-Input Multiple-Output (MIMO) system. An optimization algorithm to obtain a bit and power allocation for each subcarrier assuming instantaneous channel knowledge is used. The experimental results are shown the potential of our proposed system.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
• /
• v.9 no.6
• /
• pp.746-760
• /
• 1998

In this paper, an adaptive RSSD(reduced-state sequence detection) receiver is proposed for the purpose of reducing the complexity and decision delay of the adaptive MLSD(maximum-likelihood sequence detection) receiver in the mobile satellite channel. The RSSD receiver reconstructs the trellis with a reduced number of states. The performance degradation due to the reduced states is compensated by modifying the branch metric calculation which uses the symbols in each path memory to estimate the residual ISI(intersymbol interference) terms. The structure of the proposed adaptive RSSD is a modified RSSD utilizing a per-survivor processing as well as the symbol-aided method and a channel estimation using the tentative data sequences. The complexity and performance of the proposed adaptive RSSD are controlled by the number of system states and ISI cancelers and the inserting period of the known symbols. In spite of a suboptimal alternative receiver compared to the adaptive MLSD receiver, the proposed adaptive RSSD receiver is able to reduce the complexity significantly and track the time-varying channel fast and reliably.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.33 no.2A
• /
• pp.191-197
• /
• 2008

For the efficient transmission of burst data in the time varying wireless channel, opportunistic scheduling is one of the important techniques to maximize multiuser diversity gain. In this paper, we propose a distributed opportunistic scheduling scheme for wireless LAN network. A proportional fair scheduling, which is one of the opportunistic scheduling schemes, is used for centralized networks, whereas we design distributed proportional fair scheduling (DPFS) scheme and medium access control with distributed manner. In the proposed DPFS scheme, each receiver estimates channel condition and calculates independently its own priority with probabilistic manner, which can reduce excessive probing overhead required to gather the channel conditions of all receivers. We evaluate the proposed DPFS using extensive simulation and simulation results show that DPFS obtains higher network throughput than conventional scheduling schemes and has a flexibility to control the fairness and throughput by controlling the system parameter.