• Journal of the Institute of Electronics Engineers of Korea CI
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• v.43 no.4 s.310
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• pp.23-30
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• 2006

This paper presents design andd verification of a circuit that improves the control-operation problems of Stored Program Optical Computer (SPOC), which is an optical computer using $LiNbO_3$ optical switching element. Since the memory of SPOC takes the Delay Line Memory (DLM) architecture and instructions that are needless of operands should go though memory access stages, SPOC memory have problems; it takes immoderate access time and unnecessary operations are executed in Arithmetic Logical Unit (ALU) because desired operations can't be selectively executed. In this paper, improvement on circuit has been achieved by removing the memory access of instructions that are needless of operands by decoding instructions before locating operand. Unnecessary operations have been reduced by sending operands to some specific operational units, not to all the operational units in ALD. We show that total execution time of a program is minimized by using the Dual Instruction Register(DIR) architecture.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.40 no.6
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• pp.71-83
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• 2003

In this paper, a novel transcoding algorithm for the G.729A and the Selectable Mode Vocoder(SMV) vocoders via direct parameter transformation is proposed. In contrast to the conventional tandem transcoding algorithm, the proposed algorithm converts the parameters of one coder to the other without going through the decoding and encoding processes. In transcoder from SMV to G.729A, LSP conversion algorithm, pitch delay conversion algorithm and transcoding algorithm in lower rate are proposed, and in transcoder from G.729A to SMV, LSP conversion algorithm, pitch delay conversion algorithm and rate selection algorithm are proposed. Evaluation results show that while exhibiting better computational and delay characteristics, the proposed algorithm produces equivalent or Improved speech quality to that produced by the tandem transcoding algorithm.

• The KIPS Transactions:PartB
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• v.11B no.2
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• pp.177-186
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• 2004

It is a weak point of the motion estimation technique for video compression that the predicted video encoding algorithm requires higher-order computational complexity. To reduce the computational complexity of encoding algorithms, researchers introduced techniques such as 3D-WT that don't require motion prediction. One of the weakest points of previous 3D-WT studies is that they require too much memory for encoding and too long delay for decoding. In this paper, we propose a technique called `FS (Fast playable and Scalable) 3D-WT' This technique uses a modified Haar wavelet transform algorithm and employs improved encoding algorithm for lower memory and shorter delay requirement. We have executed some tests to compare performance of FS 3D-WT and 3D-V. FS 3D-WT has exhibited the same high compression rate and the same short processing delay as 3D-V has.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.25 no.11B
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• pp.1872-1878
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• 2000

Turbo Code는 반복 부호 알고리듬을 사용함으로써 백색 가우시안 잡음(AWGN)채널 환경하에서 Shannon의 한계에 가까운 성능을 보이는 오류정정 방식으로 제안되었다. 그러나 Turbo code는 반복복호로 인해 매 복호시마다 큰 인터리버와 복호기를 거쳐야 하기 때문에 수신과정에서 커다란 지연을 요구하게 된다. 따라서 차세대 무선 멀티미디어 통신에서 실시간 음성서비스나 화상서비스를 제공하는데 어려움이 많다. 본 논문에서는 기존의 터보 복호기를 변형하여 매 복호시 각각의 복호기에서 LLR 출력시퀀스를 발생시킴으로써 반복 복호 횟수를 줄이는 방법을 제안하였다. 이렇게함으로서 기존의 Toubo code가 갖는 성능은 크게 변화시키지 않으면서 각각의 정보프레임을 가변적으로 복호함으로서 반복 복호로 인한 시간 지연을 줄였다.

• MALSORI
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• no.52
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• pp.111-120
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• 2004

This paper focuses on the DSP implementation of an HMM-based speech recognizer that can handle several hundred words of vocabulary size as well as speaker independency. First, we develop an HMM-based speech recognition system on the PC that operates on the frame basis with parallel processing of feature extraction and Viterbi decoding to make the processing delay as small as possible. Many techniques such as linear discriminant analysis, state-based Gaussian selection, and phonetic tied mixture model are employed for reduction of computational burden and memory size. The system is then properly optimized and compiled on the TMS320C6711 DSP for real-time operation. The implemented system uses 486kbytes of memory for data and acoustic models, and 24.5 kbytes for program code. Maximum required time of 29.2 ms for processing a frame of 32 ms of speech validates real-time operation of the implemented system.

• Proceedings of the IEEK Conference
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• 2004.06a
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• pp.265-268
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• 2004

In this paper. we propose an adaptive step-size algorithm for the adaptive interference canceller (AIC) in the space-time trellis coded DS-CDMA system. In the AIC, the performance of the blind LMS algorithms that updates the tap-weight vector of the AIC is heavily dependent on the choice of step-size. To improve the performance of the fixed step-size AIC (FS-AIC), the regular adaptive step-size algorithm is extended in complex domain and applied to the joint AIC and ML decoder scheme. Simulation results show that the joint adaptive step-size AIC (AS-AIC) and ML decoder scheme using the proposed algorithm has boner performance than not only the conventional ML decoder but also the joint FS-AIC and ML decoder scheme without much increase of the decoding delay and complexity.

• Proceedings of the IEEK Conference
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• 2005.11a
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• pp.1193-1196
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• 2005

In this paper, we presents a new multimedia service transmit method using IPv6. The IPv6 provides the address system of 128 bit and the address space which is infinite it provides. But it will not become the IPv4 and interchange not to be, it uses the DSTM Transition mechanism which will reach and the IPv4 center in the packet header the service type it will be able to support the service class of multi type (TOS) it secures the weak point of data transfer delay it puts a base in the IPv6. The efficiency of this proposed technique have been proven by MPEG-4 streaming video streaming of the IPv6 namely, 6Xtream embodied a order form/live streaming server and the client which it uses to be possible in base and real-time decoding method.

• Journal of the Korean Institute of Electrical and Electronic Material Engineers
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• v.28 no.11
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• pp.748-751
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• 2015

Sound Masking System technology as by sound the same on all bands and artificially generates a constant sound shield People want to hear or recognize the people with the noise generated from the interior of the way. Prevent hearing or prevent recognition by using the technology to control the audible frequency band Continue to emit constant and uniform shielding sound audible frequency band Even the security content of speech (20 Hz~20 KHz). That interception laser eavesdropping, internal solicitations, during recording Or delay the decoding was a result of the effect of interference calculated Experience noise disturbance index is applied around the Stress Index is the average index is 10.16 was a luxury for the average index is then applied to the index 3.07 Noise is significantly lower stress level has improved noise conditions.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 1996.06a
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• pp.55-59
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• 1996

In this paper, we first analyze the influence of interference due to the time variation and delay spread of the mobile channel on an orthogonal frequency division multiplexing (OFDM) system. With the result, we obtain the bit error rate performance of the 16-QAM OFDM system. Second, we investigate the performance of the Reed-Solomon (RS) coded 16-QAM OFDM system when the number of subcarriers varies. In the investigation, we assume that the information transmission rate and the total bandwidth expansion due to coding, guard interval, and the number of subcarriers are fixed. Under this condition, it is observed that there are optimum numbers of subcarriers that minimize the post decoding symbol error probability of RS code for various channel states.

• Proceedings of the IEEK Conference
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• 2000.09a
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• pp.597-600
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• 2000

Turbo Code는 반복 복호알고리듬을 사용함으로써 백색 가우시안 잡음(AWGN)채널 환경에서 Shannon의 한계에 가까운 성능을 보이는 오류정정 방식으로 제안되었다. 그러나 Turbo code는 반복복호로 인해 매복호시마다 큰 인터리버와 복호기를 거쳐야 하기 때문에 수신과정에서 커다란 지연을 요구하게 된다. 따라서 차세대 무선 멀티미디어 통신에서 실시간으로 음성서비스나 화상서비스를 제공하는데 어려움이 많다. 본 논문에서는 기존의 터보 복호기를 변형하여 매 복호시 각각의 복호기에서 출력시퀀스를 발생시킴으로서 반복 복호 횟수를 줄이는 방법을 제안하였다. 이렇게 함으로서 기존의 Turbo code가 갖는 성능은 크게 변화시키지 않으면서 각각의 정보프레임을 가변적으로 복호함으로서 반복복호로 인한 시간 지연을 줄일수 있었다.