• Proceedings of the IEEK Conference
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• 2002.06d
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• pp.401-404
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• 2002

In general, speech coding methods are classified into the following three categories: the waveform coding, the source coding and the hybrid coding. In this paper, First, the reference waveform is detected after searching the pitch period by NAMDF similarity and similarity between the reference waveform and the waveform each pitch period. It made a decision whether the waveform is compressed with the threshold of similarity. If the waveform is compressed only magnitude and pitch information is transmitted into the input of G.723.1 vocoder. Performing through the G.723.1 vocoder, the waveform is restored with the magnitude and pitch information by PSOLA synthesis method. The result of simulation with proposed algorithm has a 31% reduction of bit rate than the standard 5.3kbps G.723.1 ACELP vocoder.

• Speech Sciences
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• v.10 no.4
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• pp.137-147
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• 2003

Speech synthesis can classify by synthesis way, that is waveform coding, source coding and mixture coding. Specially, waveform coding is suitable for high quality synthesis. However, it is not desirable by synthesis techniques of syllable or phoneme unit because it do not separate and handles excitation and formant part. Therefore, there is a need for pitch alteration method applied in synthesis by the rule in waveform coding. This study propose about pitch alteration method that use spectrum scaling after do to flatten spectra by subband linear approximation to minimize spectrum distortion. This paper show evaluation whether show excellency of some measure compared with LPC, Cepstrum, lifter function and method that propose. estimation method seeks distribution of each flattened signal and measured degree of flattened spectra Signal flattened is normalized, So that highest point amounts to zero, and distribution of signal ,whose average is zero, is calculated. this show result that measure the spectrum distortion rate to estimate performance of method that propose. The average spectrum distortion rate was kept below the average 2.12%, so the method that propose is superiors than existent method.

• Journal of Digital Convergence
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• v.11 no.7
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• pp.165-171
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• 2013

Coding patterns that appeared frequently in the source code is a typical piece of code. The functionality that difficult to modularize, such as logging or synchronization processing, and the useful sentences in programming is extracted in software as codding pattern. Large-scale software could not be analyzed fully because the number of coding pattern that can be manually investigated is limited. In this paper, the characteristics of coding patterns perform the evaluation. The goal is to extract for codding-pattern to analyzed by developer. We was selected 6 indicators and performed analysis of 4 open-source. Matrix relations between the values and characteristics of the actual pattern analysis, pattern instances, the width of the distribution of instances, the pattern repeating structure of the elements included in the rates should be analyzed for patterns and indicators that help in choosing was confirmed.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.18 no.9
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• pp.2131-2139
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• 2014

In this paper, a binary-coded physical-layer network coding (PNC) is considered when high-order modulation techniques are used at source nodes in wireless communication environments. In the conventional PNC schemes, tight power control and phase compensation are required at a relay node. However, they may not be feasible in practical wireless communication environments. Thus, we do not assume the pre-equalization in this paper, and we only utilize the channel state information at receiver (CSIR). We propose a signal detection method for the binary-coded PNC with high-order modulation, such as QPSK and 16QAM, at the source nodes, while the conventional scheme only consider the BPSK at source nodes. We also analyze the bit-error performance of the proposed technique in both uncoded and coded cases.

• Journal of Internet Computing and Services
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• v.18 no.5
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• pp.17-22
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• 2017

In a MPC coding using excitation source of voiced and unvoiced, it would be a distortion of speech waveform. This is caused by normalization of synthesis speech waveform of voiced in the process of restoration the multi-pulses of representation section. This paper present PCFBD-MPC( Position Compensation Frequency Band Division-Multi Pulse Coding ) used V/UV/S( Voiced / Unvoiced / Silence ) switching, position compensation in a multi-pulses each pitch interval and Unvoiced approximate-synthesis by using specific frequency in order to reduce distortion of synthesis waveform. Also, I was implemented that the PCFBD-MPC( Position Compensation Frequency Band Division-Multi Pulse Coding ) system and evaluate the SNRseg of PCFBD-MPC in coding condition of 8kbps. As a result, SNRseg of PCFBD-MPC was 13.4dB for female voice and 13.8dB for male voice respectively. In the future, I will study the evaluation of the sound quality of 8kbps speech coding method that simultaneously compensation the amplitude and position of multi-pulse source. These methods are expected to be applied to a method of speech coding using sound source in a low bit rate such as a cellular phone or a smart phone.

• Proceedings of the IEEK Conference
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• 1998.10a
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• pp.1247-1250
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• 1998

In speech synthesis, LF model is widely used for excitation signal for voice source coding system. But LF model does not represent the harmonic frequencies of excitation signal. We propose an effective method which use sinusoidal functions for representing the harmonics of voice source signal. The proposed method could achieve more exact voice source waveform and better synthesized speech quality than LF model.

• The Journal of the Korea institute of electronic communication sciences
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• v.6 no.4
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• pp.499-504
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• 2011

This paper propose the rule of coding for the characters and symbol, etc which are used in computer, information devices, etc. When they use the rule of coding, they may improve the efficiency of transmission in data communications by reducing the number of scrambling during the line coding in the coder in the transmitter. The paper considered the codes of one byte of 3bits column ${\times}$ 4bits rows. ASCII code is one of the representative code for the codes of one byte of 3bits column ${\times}$ 4bits rows. Therefore, this paper applied ASCII code to study the coding rule of the characters and symbols, etc.

• Journal of the Korea Institute of Information Security & Cryptology
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• v.22 no.4
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• pp.901-911
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• 2012

The removal of security vulnerabilities during the developmental stage is found to be much more effective and much more efficient than performing the application during the operational phase. The underlying security vulnerabilities in software have become the major cause of cyber security incidents. Thus, secure coding is drawing much attention for one of its abilities includes minimizing security vulnerabilities at the source code level. Removal of security vulnerabilities at the software's developmental stage is not only effective but can also be regarded as a fundamental solution. This thesis is a research about the methods of Mobile-Secure Coding Self Assessment in order to evaluate the security levels in accordance to the application of mobile secure coding of every individual, groups, and organizations.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.20 no.8
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• pp.1438-1443
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• 2016

In this paper, we investigate the effect of multiple antennas at a relay node on the performance of physical-layer network coding (PNC) in the two-way relay channel (TWRC). We assume that two source nodes have a single antenna and the relay node has multiple antennas. We extend the conventional TWRC environment with a signle antenna at both relay and source nodes to the case of multiple antennas at the relay node. In particular, we consider two decoding strategies: separate decoding (SD) and direct decoding (DD). The SD decodes each packet from the two sources and performs the network coding with bit-wise exclusive OR (XOR) operation, while the DD decodes the network-coded packet from the two sources. Note that both decoding strategies are based on log-likelihood ratio (LLR) computation. It is shown that the bit error rate (BER) performance becomes significantly improved as the number of antennas at the relay node.

• Proceedings of the Korean Institute of Intelligent Systems Conference
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• 1998.06a
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• pp.756-761
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• 1998

This paper describes an in-line lossless data compression method using adaptive binary arithmetic coding. To achieve better compression efficiency , we employ an adaptive fuzzy -tuning modeler, which uses fuzzy inference to deal with the problem of conditional probability estimation. The design is simple, fast and suitable for VLSI implementation because we adopt the table -look-up approach. As compared with the out-comes of other lossless coding schemes, our results are good and satisfactory for various types of source data.