• The Journal of Korean Institute of Communications and Information Sciences
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• v.36 no.8C
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• pp.497-504
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• 2011

This paper presents an adaptive retinex algorithm, In order to solve typical problems of retinex algorithm such as expensive computational cost, halo artifact, and color distortion, a function of skewness that represents a statistical distribution of pixels is defined to compensate contrast and color distortion. The experimental results show that the proposed algorithm leads to subjectively better performance than typical retinex algorithm, and that the proposed algorithm has the capability to reduce approximately 40% computational cost than typical retinex algorithm.

• Proceedings of the Korea Information Processing Society Conference
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• 2012.04a
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• pp.499-502
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• 2012

빅 데이터 분석의 시대가 도래하면서 대용량 데이터의 특성과 계산 집약적 연산의 특성을 동시에 가지는 문제 해결에 대한 요구가 늘어나고 있다. 대용량 데이터 처리의 경우 각종 분산 파일 시스템과 분산/병렬 컴퓨팅 기술들이 이미 많이 사용되고 있으며, 계산 집약적 연산 처리의 경우에도 GPGPU 활용 기술의 발달로 보편화되는 추세에 있다. 하지만 대용량 데이터와 계산 집약적 연산 이 두 가지 특성을 모두 가지는 문제를 처리하기 위해서는 많은 제약 사항들을 해결해야 하는데, 본 논문에서는 이에 대한 대안으로 분산 컴퓨팅 프레임워크인 Hadoop MapReduce와 Nvidia의 GPU 병렬 컴퓨팅 아키텍처인 CUDA 흘 연동하는 방안을 제시하고, 이를 밀집행렬(dense matrix) 연산에 적용했을 때 얻을 수 있는 성능 개선 효과에 대해 소개하고자 한다.

• Journal of the Institute of Electronics and Information Engineers
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• v.51 no.8
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• pp.11-19
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• 2014

In recent SONAR (sound navigation and ranging) systems, wideband active SONAR systems has received more attention than narrowband SONAR systems due to the remarkable detection performance in terms of range resolution. However, the wideband SONAR systems usually requires a huge amount of computational burden in order to achieve their own superiority. To cope with this drawback of the wideband SONAR systems, this paper proposes a fast target detection and velocity estimation method using an extended replica in wideband hyperbolic frequency modulation active SONAR system. Computer simulation shows that the proposed method can be implemented by a highly reduced computational complexity with a little performance degradation in target detection and velocity estimation compared to the conventional filter bank method.

• Journal of Navigation and Port Research
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• v.29 no.10 s.106
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• pp.877-882
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• 2005

It is necessary that Fail-Safe Seaway technology providing a continuous navigation solution though fault of navigation system is occurred in sea. This paper focus on signal processing of GPS receiver, one of receivers using the software radio technology to implement a integrated radio navigation system including satellite-based and ground-based navigation systems. It is difficult to implement the software GPS receivers using a commercial processor because of the heavy computational burden for processing the GPS signals in real time. This paper proposes an efficient multi-bit GPS signal processing scheme to reduce the computational burden for processing the GPS signals in the software GPS receiver. The proposed scheme uses a compression concept of the multi-bit replica signals and patterned look-up table method to generate the correlation value between the GPS signals and the replica signals.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.6 no.3
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• pp.445-451
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• 2002

In this paper, we propose a new tracking algorighm that achieves good tracking performance in manuevering targets while capping the computation load to "low". Kalman Filler (KF) is generally known to be poor in tracking maneuvering targets. IMM, on the other hand, compensates the weakness inherent in the mundane KF and is considered as a promising alternative for tracking maneuvering targets. However, IMM suffers from substantially increased computational load as the number of models increases. To remedy this problem, we propose a new method focused to reducing the computational load and attaining the desirable tracking performance at least as good that of IMM. It is achieved by essentially adopting the structure of IMM and injecting Optimal Two-Stage Kalman Estimator (OTSKE). The representative simulation shows a reduction in computational load with the proposed OTSKE but further reduction is shown achieved (by about 58%) with the Interacting Acceleration Compenstation (IAC)-OTSKE approach. approach.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.7C
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• pp.962-970
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• 2004

This paper presents an efficient implementation of the 2.4 kbps EHSX(Enhanced Harmonic Stochastic Excitation) speech coder on a TMS320C6701$^{TM}$ floating-point digital signal processor. The EHSX speech codec is based on a harmonic and CELP(Code Excited Linear Prediction) modeling of the excitation signal respectively according to the frame characteristic such as a voiced speech and an unvoiced speech. In this paper, we represent the optimization methods to reduce the complexity for real-time implementation. The complexity in the filtering of a CELP algorithm that is the main part for the EHSX algorithm complexity can be reduced by converting program using floating-point variable to program using fixed-point variable. We also present the efficient optimization methods including the code allocation considering a DSP architecture and the low complexity algorithm of harmonic/pitch search in encoder part. Finally, we obtained the subjective quality of MOS 3.28 from speech quality test using the PESQ(perceptual evaluation of speech quality), ITU-T Recommendation P.862 and could get a goal of realtime operation of the EHSX codec.c.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.41 no.4
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• pp.121-130
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• 2004

In the application requiring the interoperability of different networks such as VoIP and wireless communication system, two speech codecs must work together with the structure of cascaded connection, tandem. Tandem has several problems such as long delay, high complexity and quality degradation due to twice complete encoding/decoding process. Transcoding is one of the best solutions to solve these problems. Transcoding algorithm is varied with the structure of source and target coder. In this paper, transcoding algorithm including the LSP conversion, the pitch estimation and new perceptual weighting filter for reducing complexity and improving qualify is proposed. These algorithms are applied to the pair of AMR md G.723.1. By employing the proposed algorithms in the transcoder, the complexity is reduced by about 20%-58% and quality is improved compared to tandem.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.44 no.4 s.316
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• pp.109-116
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• 2007

Most harmonic synthesis methods using phase information employ a quadratic or cubic phase interpolation. The methods are computationally expensive to implement because every component sinewave must be synthesized on a per sample basis. In this paper, we propose a fast harmonic synthesis method for sinusoidal speech/audio coding based on the quadratic and cubic phase function to overcome the complexity problem. To derive the fast harmonic synthesis method, we define the over-sampling function and phase modulation function by constraining the parameter of phase function to be independent for harmonic index and derive the fast synthesis method using IFFT. Experimental results show that the proposed method significantly reduce the complexity of conventional cosine synthesis method while maintaining the performance.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.48 no.10
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• pp.53-59
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• 2011

In recent years, mobile communication service is used extensively as a larger service area for the maintenance of quality of service required by the expansion of service areas and As the ever-increasing role in relays, and the location is relatively easy to install and less constrained costs, operating cost savings in terms of ICS(Interference Cancellation System) repeaters are required. However, an adaptive algorithm that is applied when updating the filter due to the increase in volume of operations increase the complexity of hardware implementation is fraught with many difficulties. In this paper, if there is a path that feedback. ICS repeater utilizing baseband signal processing for the removal of interfering signals from the feedback operation, significantly reducing the amount of reducing hardware complexity Complex Singed Signed LMS adaption algorithm is proposed. Proposed algorithm for evaluating the performance of Static channel WCDMA signal environment for the ICS, the results of the simulation algorithm, convergence speed and better performance in therms of convergence errors that are required through the implementation of the operation greatly reduces the amount of hardware complexity able to reduce the effect was visible.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2006.11a
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• pp.111-115
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• 2006

스케일러블 비디오 코딩(SVC, Scalable Video Coding)은 MPEG(Moving Picture Expert Group)과 VCEG (Video Coding Expert Group)의 JVT(Joint VIdeo Team)에 의해 현재 표준화 되고 있는 새로운 압축 표준 기술이며 시간, 공간 및 화질의 스케일러빌리티를 지원하기 위해 계층 구조를 가지고 있다. 특히 시간적 스케일러빌리티를 위해 계층적 B-픽처 구조를 채택하고 있다. 스케일러블 비디오 코딩의 기본 계층은 H.264|AVC와 호환적이므로, 모션 예측과 모드 결정과정에서 $16{\times}16,\;16{\times}8,\;8{\times}16,\;8{\times}8,\;8{\times}4,\;4{\times}8$ 그리고 $4{\times}4$와 같은 7개의 서로 다른 크기를 갖는 블록을 사용한다. 스케일러블 비디오 코딩에서 사용되고있는 계층적 B-픽처 구조는 키 픽처인 I와 P 픽처를 제외하고는 한 GOP (Group of Picture)내에서 모두 B-픽처를 사용하므로 H.264|AVC와 비교했을 때 연산량 증가와 함께 부호화 지연도 급격히 증가한다. B-픽처는 양방향 모션 벡터인 LIST0와 LIST1을 사용하고 양방향 모두에서 다중 참조 픽처를 사용하기 때문이다. 본 논문에서는 통계적 가선 검증을 이용하여 스케일러블 비디오 부호화에 적용 가능한 고속 프레임간 모드 결정 알고리듬 대해 소개한다. 제안된 방법은 $16{\times}16$ 매크로 블록과 $8{\times}8$ 서브 매크로 블록에 통계적 가설 감증 기법을 적용하여 실행되며, 현재 블록과 복원된 참조 블록간의 픽셀 값을 비교하여 RD(Rate Distortion) 최적화 기반 모드 결정을 빨리 완료함으로써 고속 프레임간 모드 결정을 가능하게 한다. 제안된 방법은 프레임 간 모드 결정을 고속화함으로써 스케일러블 비디오 부호화기의 연산량과 복잡도를 최대 57%감소시킨다. 그러나 연산량 감소에 따른 비트율의 증가나 화질의 열화는 최대 1.74% 비트율 증가 및 0.08dB PSNR 감소로 무시할 정도로 작다.