• Proceedings of the IEEK Conference
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• 대한전자공학회 2002년도 ITC-CSCC -3
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• pp.1527-1530
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• 2002

Video summary is one of the tools which can provide the fast and effective browsing fur a lengthy video. Video summary consists of many key-frames that could be defined differently depending on the video genre it belongs to. Consequently, the video summary constructed by the uniform manner might lead into inadequate result. Therefore, identifying the video genre is the important first step in generating the meaningful video summary. We propose a new method that can classify the genre of the video data in MPEG compressed bit-stream domain. Since the proposed method operates directly on the com- pressed bit-stream without decoding the frame, it has merits such as simple calculation and short processing time. In the proposed method, only the visual information is utilized through the spatial-temporal analysis to classify the video genre. Experiments are done for 6 genres of video: Cartoon, Commercial, Music Video, News, Sports, and Talk Show. Experimental result shows more than 90% of accuracy in genre classification for the well-structured video data such as Talk Show and Sports.

• Journal of the Institute of Electronics Engineers of Korea SP
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• 제48권1호
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• pp.23-32
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• 2011

In this paper, we develop a method for generating thumbnail images at hybrid domain combined the spatial domain and transform domain. The proposed method generates a pixel of a thumbnail image by adding a DC value of residual transform coefficients and an average value of an estimate block. For effectively calculating average values of estimate blocks, we propose a method for reconstructing the boundary pixels of a block. In comparison to the conventional method of decoding the bit stream then scaling down the decoded images, the developed method reduces the complexity by more than 60% while producing identical thumbnail images.

• Journal of the Korea Institute of Information and Communication Engineering
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• 제18권12호
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• pp.2871-2877
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• 2014

Real-time communication and diversification of multimedia application services using high-capacity data over the Internet demand high speed connection, and real-time and high-speed access internet service such as multimedia make Internet traffic increase rapidly. Meanwhile, QoS (Quality of Service) of internet service is not satisfied. In this situation, ISP (Internet Service Provider) has been required to improve service quality and extend network according to the user's needs. This request includes issue of extending network focused on increasing router and the number of routing table as well as simply extending bandwidth.

• KSII Transactions on Internet and Information Systems (TIIS)
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• 제12권7호
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• pp.3455-3474
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• 2018

As a unique form of signed-digit representation, non-adjacent form (NAF) minimizes Hamming weight by removing a stream of non-zero bits from the binary representation of positive integer. Thanks to this strong point, NAF has been used in various applications such as cryptography, packet filtering and so on. In this paper, to improve the NAF conversion speed of the $NAF_w$ algorithm, we propose a new NAF conversion algorithm, called w-bit Shifting Non-Adjacent Form($SNAF_w$), where w is width of scanning window. By skipping some unnecessary bit comparisons, the proposed algorithm improves the NAF conversion speed of the $NAF_w$ algorithm. To verify the excellence of the $SNAF_w$ algorithm, the $NAF_w$ algorithm and the $SNAF_w$ algorithm are implemented in the 8-bit microprocessor ATmega128. By measuring CPU cycle counter for the NAF conversion under various input patterns, we show that the $SNAF_2$ algorithm not only increases the NAF conversion speed by 24% on average but also reduces deviation in the NAF conversion time for each input pattern by 36%, compared to the $NAF_2$ algorithm. In addition, we show that $SNAF_w$ algorithm is always faster than $NAF_w$ algorithm, regardless of the size of w.

• Journal of the Korean Institute of Telematics and Electronics
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• 제27권4호
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• pp.491-501
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• 1990

Recently, four speech coding techniques, namely, SBC-APCM(sub-band coding adaptive PCM), RPE-LPC(regualr pulse excitation linear predictive codec), MPE-LTP(multi-pulse excited long-term prediction) and CELP (code-excited linear prediction) are proposed for digital mobile radio applications. However, a performance comparison of these coders in the Rayleigh fading environment has not been made yet. In this paper, the performances of the four spech coders in the random bit error and burst error environment are investigated. For the channel coding of SBC-APCM, RPE-LPC and MPE-LTP, the sensitivity of output bit stream is measured and a bit selective forward error correction is provided acording to the measured bit sensitivity. And for an attempt to improve the performance of CELP, an optimum quantizer is applied for transmitting scalar quantities in CELP. However, an improvement over the conventional approach is found to be negligible. For the channel coding of CELP, Reed-Solomon code, Golay code, convolutional code of rate 1/2 shows the best performance. Finally, from the simulation results, it is concluded that CELP is the best candidate for digital mobile radio and is followed by MPE-LTP, SBC-APCM and RPE-LPC.

• Journal of the Institute of Convergence Signal Processing
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• 제3권3호
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• pp.14-20
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• 2002

In this paper, we proposed the compression technique of the video data using SPIHT(set partitioning in hierarchical trees) based on wavelet and the multi-stream delivery method for best-effort service as fully utilizing the clients bandwidth over the current Internet. The experiment shows that the proposed method provides about 1.5dB better picture quality without block effects than DCT(discrete consine transform) based coding schemes at the same bit rates because of using the wavelet video coder. In addition, this technique implements the multi-stream transmission based on TCP(transmission control protocol). Thus, it is provided with the best-efforts service which is robust to the network jitter problem, and maximally utilizes the bandwidth of the client's.

• The Journal of Korean Institute of Communications and Information Sciences
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• 제38C권11호
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• pp.972-981
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• 2013

Future aspects of the has turned into a network centric warfare(NCW). Organically combined wired and wireless networks in a variety of cross-of-the-art combat power factor utilization of information and communication technology is a key element of NCW implementation. At used various information in the NCW must be the confidentiality and integrity excellent then quick situation assessment through reliability the real-time processing, which is the core of winning the war. In this paper, NCW is one of the key technologies of the implementation of 128-bit output stream cipher algorithm is proposed. AES-based stream cipher developed by applying modified OFB mode the confidentiality and integrity as well as hardware implementation to the security and real-time processing is superior.

• Journal of the Korea Institute of Information and Communication Engineering
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• 제14권4호
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• pp.859-867
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• 2010

In this paper, we propose some parallel structures of the word-based nonlinear combining functions in word-based stream cipher, high-speed versions of general (bit-based) nonlinear combining functions. Especially, we propose the high-speed structures of popular four kinds in word-based nonlinear combiners using by PS-WFSR (Parallel-Shifting or Parallel-Structured Word-based FSR): m-parallel word-based nonlinear combiner without memory, m-parallel word-based nonlinear combiner with memories, m-parallel word-based nonlinear filter function, and m-parallel word-based clock-controlled function. In addition, we propose an implementation example of the m-parallel word-based DRAGON stream cipher, and determine its cryptographic security and performance.

• Journal of Broadcast Engineering
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• 제2권1호
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• pp.8-15
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• 1997

In this paper, a real-time operating Motion Picture Experts Group-1 (MPEG-1) audio encoder system is implemented using a TMS320C31 Digital Signal Processor (DSP) chip. The basic operation of the MPEG-1 audio encoder algorithm based on audio layer-2 and psychoacoustic model-1 is first verified by C-language. It is then realized using the Texas Instruments (Tl) assembly in order to reduce the overall execution time. Finally, the actual BSP circuit board for the encoder system is designed and implemented. In the system, the side-modules such as the analog-to-digital converter (ADC) control, the input/output (I/O) control, the bit-stream transmission from the DSP board to the PC and so on, are utilized with a field programmable gate array (FPGA) using very high speed hardware description language (VHDL) codes. The complete encoder system is able to process the stereo audio signal in real-time at the sampling frequency 48 kHz, and produces the encoded bit-stream with the bit-rate 192 kbps. The real-time operation capability of the encoder system and the good quality of the decoded sound are also confirmed using various types of actual stereo audio signals.

• The Journal of Korean Institute of Communications and Information Sciences
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• 제29권8C호
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• pp.1210-1217
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• 2004

LILI-II stream cipher is an upgraded version of the LILI-128, one of candidates in NESSIE. Since the algorithm is a clock-controlled, the speed of the keystream data is degraded structurally in a clock-synchronized hardware logic design. Accordingly, this paper proposes a 4-bit parallel LFSR, where each register bit includes four variable data routines for feedback or shifting within the LFSR. furthermore, the timing of the proposed design is simulated using a Max+plus II from the ALTERA Co., the logic circuit is implemented for an FPGA device (EPF10K20RC240-3), and apply to the Lucent ASIC device (LV160C, 0.13${\mu}{\textrm}{m}$ CMOS ＆ 1.5v technology), and it could achieve a throughput of about 500 Mbps with a 0.13${\mu}{\textrm}{m}$ semiconductor for the maximum path delay below 1.8㎱. Finally, we propose the m-parallel implementation of LILI-II, throughput with 4, 8 or 16 Gbps (m=8, 16 or 32).