• The Journal of Korean Institute of Communications and Information Sciences
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• v.24 no.9B
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• pp.1721-1730
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• 1999

In this thesis, an error resilient video coding algorithm, under the error-prone environment such as wireless communication, is suggested. The suggested algorithm adapts the Classified VQ method for intra imagers that reduces some load by searching similar vectors. The Duplicate Vector Position Code is proposed for the higher compression efficiency and the robust decoding in error environment. As a result, the bitstream encoded by the proposed method is in a CBR(Constant Bit Rate) preventing from error propagation. The experiment that adds practical error to the encoded bitsrteam shows the error-robustness superior to H.263.

• The KIPS Transactions:PartC
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• v.13C no.1 s.104
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• pp.75-82
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• 2006

In this paper, a space-frequency coded orthogonal frequency division multiplexing (SFC-OFDM) scheme with dual Viterbi decoder is proposed and analyzed. Here, two independent half-rate OFDM symbols are generated after convolutional coding of the binary source code. A dual Viterbi decoder is exploited to decode the demodulated sequences independently in the receiver, and their path metrics are compared. Accordingly, the recovered binary data in the proposed scheme are composed of the combination of the sequences having larger path metrics while those in a conventional system are simply the output of single Viterbi decoder. As a result, the proposed SFC-OFDM scheme has a better performance than the conventional one for all signal-to-noise power ratio.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.10 no.1
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• pp.39-50
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• 2010

In this paper, we analyze and simulate the performance of a tactics mobile communication system based on ultra wide band (UWB) in multi-user interference (MUI) environments. This system adopts a double binary turbo code for forward error correction (FEC). Wireless channel is modeled a modified Saleh and Valenzuela (SV) model. We employ a space time block coding (STBC) scheme for enhancing system performance. System performance is evaluated in terms of bit error probability. From the simulation results, it is confirmed that the tactics mobile communication system based on UWB, which is encoded with the double binary turbo code, can achieve a remarkable coding gain with reasonable encoding and decoding complexity in multi-user interference environments. It is also known that the bit error probability performance of the tactics mobile communication system based on UWB can be substantially improved by increasing the number of iterations in the decoding process for a fixed cod rate. Besides, we can demonstrate that the double binary turbo coding scheme is very effective for increasing the number of simultaneous users for a given bit error probability requirement.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2012.11a
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• pp.101-102
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• 2012

지수-골롬 부호는 데이터의 무손실 부호화를 위해 사용되는 대표적인 방법으로서 지수 함수 형태로 단일 모드 분포를 가지는 데이터의 부호화에 적합하다. 본 논문에서는 데이터의 분포가 단일 모드가 아닌 이중 모드를 가지는 경우에 적용하기 위해 변형된 지수-골롬 부호를 제안하고 모드의 높이와 모드 사이의 거리에 따른 부호화 성능을 살펴보았으며 실제 데이터에 적용한 실험결과를 제시하였다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.5C
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• pp.548-554
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• 2003

Interoperability is ole the most important factors for a successful integration of the speech network. To accomplish communication between endpoints employing different speech coders, decoder and encoder of each endpoint coder should be placed in tandem. However, tandem coder often produces problems such as poor speech quality, high computational load, and additional transmission delay. In this paper, we propose an efficient transcoding algorithm that can provide interoperability to the networks employing ITU-T G.723.1［1］and TIA IS-127 EVRC［2］speech coders. The proposed transcoding algorithm is composed of four parts: LSP conversion, open-loop pitch conversion, fast adaptive codebook search, and fast fixed codebook search. Subjective and objective quality evaluation confirmed that the speech quality produced by the proposed transcoding algorithm was equivalent to, or better than the tandem coding, while it had shorter processing delay and less computational complexity, which is certified implementing on TMS320C62x.

• Proceedings of the Acoustical Society of Korea Conference
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• 1998.06e
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• pp.415.1-418
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• 1998

G.723.1은 부호화 방식은 제한된 계산량으로 낮은 전송율에서 음성을 가장 잘 표현할 수 있도록 최적화되어 있어서, 음성주파수 대역에 있는 DTMF톤의 경우 왜곡이 발생되어 전송성능이 떨어지는 문제점이 있다. 본 논문에서는 DTMF톤의 투명한 전송을 위해 LSP 계수를 이용한 톤 신호 검출에 기반을 둔 음성모드와 톤모드의 이중모드를 가지는 변형된 G.723.1 음성부호화 방식을 제안한다. 제안된 방식에서 음성모드 부호화기는 기존의 방식과 동일하며, 톤모드의 경우 부호화 단계에서 spectral smoothing 및 피치주기 검출 방식 등을 수정함으로써 수신단의 변경없이 DTMF톤의 전송왜곡을 개선시킨다. 본 논문에서는 컴퓨터 모의실험을 통해 제안된 방법이 DTMF 전송성능을 개선시킴을 확인하였다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.26 no.10B
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• pp.1399-1408
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• 2001

본 논문에서는 삼각형 메쉬(triangular mesh)로 이루어진 3차원 모델의 연결 정보(connectivity data)와 형상 정보(geometry data)를 삼각형 팬(triangle fan) 구조에 기반하여 효율적으로 압축하는 부호화 기법을 제안한다. 첫째로, 연결 정보의 무손실 부호화로 꼭지점 차수 왜곡(vertex degree warping) 기법을 제안한다. 기존의 알고리듬이 연결 정보와 형상 정보를 분리하여 부호화하는데 반해 제안하는 알고리듬은 연결 정보를 부호화하는데 형상 정보를 이용하여 압축 효율을 높인다. 둘째로, 형상 정보를 압축하기 위해 이중 평행사변형 예측(dual parallelogram prediction) 기법을 제안한다. 삼각형 팬 구조를 이용함으로써 기존의 삼각형 스트립(triangle strip) 기반의 알고리듬보다 정확한 형상 예측이 가능하고, 예측 오차가 작아지게 된다. 다양한 3차원 메쉬 모델에 대한 모의 실험을 통하여 제안하는 알고리듬이 기존의 알고리듬보다 우수한 압축 성능을 나타냄을 확인하다.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.45 no.1
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• pp.51-58
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• 2008

We propose a new duplex which does not require guard resources in MIMO environments. In the proposed duplex, two virtual channels in spatial domain are generated by preceding and postcoding MIMO channels, not to use guard resources in either the time or frequency domain. The capacity can be improved by properly selecting precoder and postcoder. We show the improved capacity of the proposed duplex outperforms conventional duplexes even without guard resources.

• Journal of Broadcast Engineering
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• v.23 no.1
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• pp.86-92
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• 2018

In this paper, we propose a method for coding history detection of digital speech signal. In digital speech communication and storage, the signal is encoded to reduce the number of bits. Therefore, when a speech signal waveform is given, we need to detect its coding history so that we can determine whether the signal is an original or an coded one, and if coded, determine the number of times of coding. In this paper, we propose a coding history detection method for 12.2kbps AMR codec in terms of original, single coding, and double coding. The proposed method extracts a speech-specific feature vector from the given speech, and models the feature vector using a deep neural network. We confirm that the proposed feature vector provides better performance in coding history detection than the feature vector computed from the general spectrogram.

• Journal of the Korean Institute of Telematics and Electronics
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• v.21 no.6
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• pp.115-117
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• 1984

A method of calculating the maximum entropy per run in binary run-length coding is given when run-length is limited to a maximum of M. And Huang's bounds can be obtained from the present result as M tends to infinity.