• Journal of the Korean Institute of Telematics and Electronics C
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• v.36C no.1
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• pp.1-11
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• 1999

In this paper, we proposed a design method of parameterized viterbi decoder, which automatically synthsizes the diverse viterbi deciders used in the digital mobile communication systems. It is designed to synthesize a viterbi decoder specified by user-provided parameters. Those parameters are constraint length, code rate generator polynomials of teh convolutional encoder, data rate and bits/frame of the data transmission, and soft decision bits of viterbi decoder. For the design of the parameterized viterbi decoder, we designed a user interface module C-language, and a viterbi decoder module in a hierarchical atructure using VHDL language and its generic statement. For the verification of the parameterized viterbi decoder, we compared our synthesized viterbi decoder with the conventional viterbi decoder which is designed for the IS-95 CDMA system. The proposed design method of the viterbi decoder will be a new method to obtain a required viterbi decoder in a very short time only by supplying the design parameters.

• The KIPS Transactions:PartC
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• v.14C no.3 s.113
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• pp.285-292
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• 2007

In the adaptable bandwidth allocation technique, a transmission plan for variable rate video data is made by the CBA algorithm and the data is transmitted by considering network traffic. But the CBA algorithm produces a transmission plan where the size of the increasing interval of transmission rate is generally larger than the size of the decreasing interval. And the transmission rate in CBA algorithm is changed in overflow curve during the increasing interval of transmission rate. This ぉy cause many frames to be discarded when available transmission rate is small. In this paper, a smoothing algorithm is proposed, where transmission rate is changed in the middle of underflow curve and overflow curve, but the transmission rate increases at the minimum. In order to show the performance, the proposed algorithm and a CBA algorithm were applied to a transmission plan in the adaptable bandwidth allocation technique, and the minimum frame rates, the average frame rates, the variation of frame rates, and the numbers of discarded frames were compared in both algorithms.

• Journal of Internet of Things and Convergence
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• v.9 no.2
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• pp.11-17
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• 2023

Smoothing is to creating a transmission plan consisting of sections of frames that can be sent at the same transmission rate for compressed and stored video data. Various algorithms have been studied for the smoothing to minimize the number of transmission rate changes, the number of transmission rate changes, and the amount of transmission rate increase. This study evaluates the performance of a smoothing algorithm that minimizes the increase in transmission rates and maximizes the increase in transmission rates when the transmission rate is required to maximize the excess bandwidth to be secured by the server in an environment with limited server bandwidth. The available transmission rates and buffer sizes available in the server are set in various ways and evaluated by the number of fps changes, the minimum fps, the average fps, and fps variability. As a result of the comparison, the proposed algorithm showed excellent average fps and fps variability.

• The Journal of the Korea institute of electronic communication sciences
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• v.10 no.11
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• pp.1225-1230
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• 2015

MAC protocol for WBAN performs CSMA/CA(:Carrier Sense Multiple Access with Collision Avoidance)algorithm that handles traffic which occurs in emergency with top priority in order to deal with variable data of sensor node for medicine. Emergency message that node sends undergoes collision and delay of transmission by retransmission of emergency message and waste of energy by retransmission. This paper suggests algorithm that minimizes loss of frame caused by collision and applies different backoff parameters by setting order of priority between MAC instruction frame and data frame in CSMA/CA algorithm case which applies order of priority queuing to improve aforementioned problem. According to performance evaluation, it has been found that using suggested MAC protocol showed lower collision probability, higher packet transmission processing ratio and lower packet loss compared with using IEEE 802.15.6.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.26 no.11C
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• pp.64-74
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• 2001

In this paper, a dynamic minislot reservation protocol(DMRP) is proposed and analyzed to improve the performance of the multicast traffic processing in wavelength division multiplexing(WDM) networks. For the proposed protocol analysis, the WDM network architecture with a passive star coupler is used. One pair of transceiver is used for a control channel and the ocher pair of transceiver with one fixed transmitter and one tunable receiver are used for data transmission. In this protocol. the packets which fail to have successful transmission in the present time slot due to data channel and destination collisions, have priority to have successful reservation in the contention-less minislots of the next time slot. Therefore, protocols have improved the throughput and the system delay performance caused by the reduced probabilities of control channel contentions and destination collisions probability. Today, the efficient protocol that can handle the various types of data traffic is needed. As results, the DMRP is useful to improve the performance of throughput and system delay characteristics regardless of the kinds of traffic.

• The KIPS Transactions:PartB
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• v.11B no.6
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• pp.625-632
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• 2004

Since the H.264 encoding adopts both forward prediction and hi-direction prediction modes and exploits Variable Length Coding(VLC), the amount of data generated from video encoder varies as Flaying time goes by. The fixed bit rate encoding system which has limited transmission channel capacity uses a buffer to control output bitstream It's necessary to control the bitstream to maintain within manageable range so as to protect buffer from overflow or underflow. With existing bit amount control algorithms, the $\lambda_{MODE}$ which is relationship between distortion value and quantization parameter often excesses normal value to end up with video error. This paper proposes an algorithm to protect buffer from overflow or underflow by introducing a new quantization parameter against distortion value of H.264 video data. The test results of 6 exemplary data show that the proposed algorithm has the same PSNR as and up to 8% reduced bit rate against existing algorithms.

• Journal of Broadcast Engineering
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• v.24 no.1
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• pp.118-131
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• 2019

In order to provide 8K UHD contents of terrestrial broadcasting with a large capacity, the terrestrial broadcasting system has various problems such as limited bandwidth and so on. To solve these problems, UHD contents transmission technology has been actively studied, and an 8K UHD broadcasting system using terrestrial broadcasting network and communication network has been proposed. The proposed technique is to solve the limited bandwidth problem of terrestrial broadcasting network by segmenting 8K UHD contents and transmitting them to heterogeneous networks through hierarchical separation. Through the terrestrial broadcasting network, the base layer corresponding to FHD and the additional enhancement layer data for 4K UHD are transmitted, and the additional enhancement layer data corresponding to 8K UHD is transmitted through the communication network. When 8K UHD contents are provided in such a way, user can receive up to 4K UHD broadcasting by terrestrial channels, and also can receive up to 8K UHD additional communication networks. However, in order to transmit the 4K UHD contents within the allocated bit rate of the domestic terrestrial UHD broadcasting, the compression rate is increased, so a certain level of image deterioration occurs inevitably. Due to the nature of UHD contents, video quality should be considered as a top priority over other factors, so that video quality should be guaranteed even within a limited bit rate. This requires packet scheduling of content generators in the broadcasting system. Since the multiplexer sends out the packets received from the content generator in order, it is very important to make the transmission time and the transmission rate of the process from the content generator to the multiplexer constant and accurate. Therefore, we propose a variable transmission scheduler between the content generator and the multiplexer to guarantee the image quality of a certain level of UHD contents in this paper.

• Journal of the Institute of Electronics and Information Engineers
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• v.51 no.4
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• pp.76-89
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• 2014

In this paper, we propose a MAC(Media Access Control) protocol based on flexible RWT(RTS Waiting Time) for underwater mobile ad-hoc networks with a three-way handshaking mechanism. This protocol can solve a problem of collision between RTS(Request-To-Send) and CTS(Clear-To-Send) packets in existing MACA(Multiple Access with Collision Avoidance) protocol. This proposed MAC protocol is also an effective protocol which can apply to underwater mobile ad-hoc networks in a real field by using implementable technologies. We set flexible RTS Waiting Time called RWT, considering various characteristics of underwater environment. It is possible to support variable network size according to node mobility. Finally, we conduct a performance evaluation between proposed MAC protocol and existing MACA based MAC protocol through practical implementation and experiment. As a result, we verify the superiority of our proposed MAC protocol in terms of throughput, packet drop rate, average transmission time, energy consumption and channel utilization.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.25 no.8A
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• pp.1238-1246
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• 2000

In a multi-channel network based on wavelength division multiplexing (WDM), an efficient protocol is needed for transmitter and receiver to be tuned to same wavelength during message transmission time. This paper proposes a message scheduling protocol that can efficiently support variable-sized messages, where tunable transceiver has nonzero tuning time. In this protocol, once a node reserves one data channel, it can persistently use the reserved channel till message transmission is finished. Therefore, overhead due to tuning time of the tunable transceiver can be reduced. Moreover, the protocol does not require any global information. Therefore, it can operate independently of the change of the number of nodes, and any new node can join the network at anytime without requiring network initialization. With this protocol, one can avoid data channel and destination conflicts. The protocol is analyzed with a finite population model and the throughput-delay characteristics are investigated as performance measures.

• The Transactions of The Korean Institute of Electrical Engineers
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• v.62 no.4
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• pp.562-565
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• 2013

A new variable rate speech coding technique is proposed. The method is based on the observation that the speech signal approximately looks linear for a very short period of time. The information transmitted is the location and data value of inflection points. If the distance between the inflection points is large, the mid point location and its data value are also delivered. Thus, the encoder transmits both the location and the data value for the inflection samples, but the location only for the non-inflection points. The location information is expressed using one bit for each sample, 0 for non-inflection and 1 for inflection point. At the receiver, using the interpolation, the decoder estimates the untransmitted sample values for non-inflection locations from the received sample values for the inflection samples. With 50 % of computational cost of the existing CVSD delta modulation, the proposed method is expected to achieve the data rate of 36 to 38 kbps and the SNR of 10 to 13 dB.