• MALSORI
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• no.44
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• pp.73-82
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• 2002

Channel distortion degrades the performance of speech recognizer in telephone environment. It mainly results from the bandwidth limitation and variation of transmission channel. Variation of channel characteristics is usually represented as baseline shift in the cepstrum domain. Thus undesirable effect of the channel variation can be removed by subtracting the mean from the cepstrum. In this paper, to improve the recognition performance of Korea connected digit telephone speech, channel compensation methods such as CMN (Cepstral Mean Normalization), RTCN (Real Time Cepatral Normalization), MCMN (Modified CMN) and MRTCN (Modified RTCN) are applied to the static MFCC. Both MCMN and MRTCN are obtained from the CMN and RTCN, respectively, using variance normalization in the cepstrum domain. Using HTK v3.1 system, recognition experiments are performed for Korean connected digit telephone speech database released by SITEC (Speech Information Technology & Industry Promotion Center). Experiments have shown that MRTCN gives the best result with recognition rate of 90.11% for connected digit. This corresponds to the performance improvement over MFCC alone by 1.72%, i.e, error reduction rate of 14.82%.

• Journal of the Korean Operations Research and Management Science Society
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• v.24 no.3
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• pp.63-72
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• 1999

Recent studies on the Constant Bit Rate and Variable Bit Rate transmissions have mainly focused on the frame by frame encoder rate control based on the quantization parameter. With the existing approaches it is difficult to guarantee a consistent video quality. Also, the rate control overhead is too high for the real-time video sources. In this paper, a channel rate allocation scheme based on the control period is proposed to transmit a real-time video, in which the control period is defined by a pre-specified number of frames or group of pictures. At each control period, video traffic information is collected to determine the channel rate at the next control period. The channel rate is allocated to satisfy various channel rate constraints such that the buffer occupancy at the decoder is maintained at a target level. If the allocated channel rate approaches the level at which the negotiated traffic descriptions may be violated, the encoder rate is decreased through adjusting quantization parameters in the MPEG encoder. In the experimental results, the video quality and the overflow and underflow probabilities at the buffer are compared at different control periods. Experiments show that the video quality and the utilization of network bandwidth resources can be optimized through the suitable selection of the control period.

• Proceedings of the Korean Operations and Management Science Society Conference
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• 2005.10a
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• pp.321-333
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• 2005

With the limited frequency spectrum and an increasing demand for cellular communication services, the problem of channel assignment becomes increasingly important. However, finding a conflict free channel assignment with the minimum channel span is NP hard. As demand for services has expanded in the cellular segment, sever innovations have been made in order to increase the utilization of bandwidth. The innovations are cellular concept, dynamic channel assignment and hierarchical network design. Hierarchical network design holds the public eye because of increasing demand and quality of service to mobile users. We consider the frequency assignment problem and the base station placement simultaneously. Our model takes the candidate locations emanating from this process and the cost of assigning a frequency, operating and maintaining equipment as an input. In addition, we know the avenue and demand as an assumption. We propose the network about the profit maximization. This study can apply to GSM(Global System for Mobile Communication) which has 70% portion in the world. Hierarchical network design using GA(Genetic Algorithm) is the first three-tier (Macro, Micro, Pico) model, We increase the reality through applying to EMC (Electromagnetic Compatibility Constraints). Computational experiments on 72 problem instances which have 15${\sim}$40 candidate locations demonstrate the computational viability of our procedure. The result of experiments increases the reality and covers more than 90% of the demand.

• The Journal of the Acoustical Society of Korea
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• v.32 no.5
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• pp.377-384
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• 2013

In this paper, performance of COFDM (Coded Orthogonal Frequency Division Multiplexing) which is OFDM with a forward error correction code, is studied in frequency selective fading underwater acoustic communication channel. The OFDM is a multiplexing technique resistant to frequency selective multipath channel. In OFDM, a broadband information signal is transformed into several narrow band signals and transmits narrow band signals whose bandwidths are less than the channel coherence bandwidth. However, its performance is degraded in a specific narrow band signal due to its deep fading by multipath. To mitigate this degradation, COFDM which is OFDM with convolution code as a forward error correction code, is evaluated. The performance of COFDM is found to be better than that of OFDM in multipath channel.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.14 no.6
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• pp.729-743
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• 1989

A spread spectrum(ss) communication system has a lot of advantage, such as realzation of asynchronous code division multiplexing(CDM), robutness to narrow-band interference, impulse noise, privacy function, and so on. We have considered utilization of these advantage to develop a local area network(LAN) using such a transmission channel as power-line. In a power-line, however, restricted bandwidth under the law makes it difficult to carry out CDM, and in a radio channel the sysstem has a near-far problem. These problems originate in co-channel interference, which is due to cross-correlation among pseudo-noise sequences of every channel in CDM. In this paper, we propose a new scheme of easily cancelling co-channel interference and investigate its performance by constructing a LAN using SS technique.

• Journal of Korea Multimedia Society
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• v.11 no.5
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• pp.685-691
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• 2008

Recently with the rapid evolution of the mobile computing and communication technologies, mobile VOD service becomes increasingly important for wireless mobile users. The VOD service is being widely used in various areas of application, such as education, entertainment and business, because it provides users convenience in easily having access to video information at any time in any places. However, in reality, the mobile system has many difficulties in providing the smooth VOD service owing to frequent transfers and cutoffs of clients. The importance of a technique to transmit broadcasting is being stressed as a method for providing stabler mobile VOD service to a large number of clients. This paper is aimed at showing how to reduce demands for server bandwidth and delay of earlier service through performance analysis by suggesting an effective VOD broadcasting transmission technique through channel division in the mobile atmosphere. Many researches have been made about regular broadcasting techniques in particular. This study divides the methods used for assigning channels which have been decided by the size of segments into a group of regular channels and assistant channels using wireless gap-fillers to provide effective VOD services to a large number of clients at the mobile environment using small bandwidth resources. The regular channels transfer regular streams, while assistant channels repeatedly transfer the first segment to reduce early service delay time to receive regular streams. In this way, the study suggests a technique to reduce server bandwidth demand and early service delay time. Through the proposed technique, the server bandwidth demand could be reduced by more than 30 percent and the study continuously shows reduced early service delay time through conducting performance analysis.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.12 no.5
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• pp.2305-2310
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• 2011

Recently, communication industry based on data and voice and broadcasting industry centering around images have been rapidly blended. Thereupon, this article aims to suggest a multi-approach method which minimizes the use of network bandwidth allowing multimedia streaming transmission to secure IP-based continuity and let users get multimedia services of one channel or several simultaneously. Also, this study intends to design a buffering strategy that can absorb network delay and an object model to assign and maintain stable channel bandwidth.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.5 no.7
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• pp.1326-1331
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• 2001

This paper has been studied about design of a transceiver for data transmission. The transceiver has bandwidth of 424.7~424.95 MHz and uses half duplex communication method, PLL synthesized, 20 channel, 12.5 KHz channel bandwidth and FSK modulation/demodulation method. The transmission set is designed using low noise amplifier and power amplifier Also, it consists of low pass filter and resonation circuit for decrease of spurious signal. The receiver set is designed using dual conversion method. Finally, the transceiver set achieves the following characteristics 9.71dbm output power, 47dbc spurious property and $\pm$12.3 Jitter at sensitivity of -1134dbm.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.9 no.9
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• pp.3594-3610
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• 2015

Recent advancements in broadband wireless communication technologies enable mobile users to receive video streaming services with various smart devices. The long term evolution (LTE) network provides high bandwidth and low latency for several emerging mobile applications. This paper proposes the buffer aware scheduling (BAS) approach to schedule the downlink video traffic in LTE network. The proposed BAS scheme applies the weighting function to heuristically adjust the scheduling priority by considering the buffer status and channel condition of UE so as to reduce the time that UE stays in the connected state without receiving data. Both of 1080P and 2160P resolution video streaming sources were applied for exhaustive simulations to examine the performance of the proposed scheme by comparing to that of the fair bandwidth (FB) and the best channel quality indicator (CQI) schemes. The simulation results indicate that the proposed BAS scheme not only achieves better performance in power saving, streaming delivery time, and throughput than the FB scheme while maintaining the similar performance as the best CQI scheme in light traffic load. Specifically, the proposed scheme reduces streaming delivery time and generates less signaling overhead than the best CQI scheme when the traffic load is heavy.

• Journal of the Optical Society of Korea
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• v.13 no.3
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• pp.354-360
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• 2009

We investigate theoretically the tolerance of 40 Gbps phase-modulated (PM) duobinary signals using a vestigial sideband (VSB) filter on impairments which occurred in dense wavelength-division multiplexing (DWDM) systems, compared to the conventional duobinary signals. Our simulation results show that PM duobinary signals can't have the gain on the spectral efficiency achieved by utilizing the VSB filtering technique. In order to increase the spectral efficiency, they indispensably require to be transmitted at the optimum bandwidth of multiplexer (MUX) and demultiplexer (DEMUX) since they are susceptible to inter-channel crosstalk. It is also shown that the PM duobinary modulation format has a large tolerance on self-phase modulation (SPM) and cross-phase modulation (XPM) under the condition which MUX and DEMUX have been tuned at an optimum bandwidth; it has 1.2 dB power penalty at the fiber launching power (FLP) of 15 dBm and the channel spacing of 50 GHz.