• Proceedings of the IEEK Conference
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• 2003.07e
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• pp.2236-2239
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• 2003

본 논문에서는 G.729A 와 SMV 음성 부호화기를 위한 새로운 파라미터 직접 변환 방식의 상호 부호화 알고리즘을 제안한다. 상호 부호화를 위하여 부가적인 복호화, 부호화 과정을 거쳐야하는 기존의 tandem 방식과 달리 제안된 파라미터 직접 변환 방식에서는 양 음성부호화기에서 공통적으로 사용하는 파라미터들이 직접 변환된다. SMV에서 G.729A로의 상호 부호화에서는 LSP 변환, 피치 지연 변환, 낮은 전송률에서의 상호 부호화 둥의 알고리즘을 제안하고, G.729A에서 SMV로의 상호 부호화에서는 LSP 변환, 피치 지연 변환, 전송률 결정 등의 알고리즘을 제안한다. 제안된 알고리즘을 다양한 방법으로 평가해본 결과 기존의 tandem 방식과 비교하여 계산량과 지연 시간을 줄이면서도 동등한 음질 또는 향상된 음질을 구현함을 확인할 수 있었다.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.11 no.6
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• pp.57-62
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• 2011

In this paper an efficient LT decoding scheme using GE triangularization is proposed. The proposed algorithm has the desirable performance in terms of both overhead and computational complexity. Belief propagation algorithm is a fast and simple decoding scheme for LT codes. However, for a small code block length k, it requires a large overhead to decode, and OFG which has a small overhead has a large computational complexity. Simulation results show that the proposed algorithm noticeably reduces the computational complexity by more than 1/5 with respect to that of OFG and also its overhead has a small value about 1~5%.

• The Journal of the Acoustical Society of Korea
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• v.20 no.7
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• pp.63-68
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• 2001

This paper describes the details of ITU-T SGIS G.729A speech coder implementation using ARM9 Thumb/sup R/ processor core and various techniques used in the optimization process. ITU-T G.729 speech coder is the standard of the toll quality 8 kbit/s speech coding. The input to the speech encoder is assumed to be a 16 bits PCM signal at a sampling rate of 8000 samples per second. G.729A is reduced complexity version of the G.729 coder. This version is bit stream interoperable with the full version. The implemented coder requires 34.8 MIPS for the encoder and 8.1 MIPS for the decoder, 36.5 kBytes of program ROM and 6.3 kBytes of data RAM, respectively. The implemented coder is tested against the set of 9 test vectors provided by ITU-T for bit exact implementation.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.3C
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• pp.239-247
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• 2003

This paper presents a dual-rate (2.4/4.0 kbps) Improved Harmonic-CELP(IHC) speech coder based on the EHC(Efficient Harmonic-CELP) which was presented by the authors. The proposed IHC employs the harmonic coding for voiced and the CELP for unvoiced segments. In the IHC, an initial voiced/unvoiced estimate is obtained by the pitch gain and energy. Then, the final V/UV mode is decided by using the frame energy contour. A new harmonic estimation combining peak picking and delta adjustment provides a more reliable harmonic estimation than that in the EHC. In addition, a noise mixing scheme in conjunction with an improved band voicing measurement provides the naturalness of the synthesized speech. To demonstrate the performance of the proposed IHC coder, the coder has been implemented and compared with the 2.0/4.0 kbps HVXC(Harmonic excitation Vector Coding) standardized by MPEG-4. Results of subjective evaluation showed that the proposed IHC coder and produce better speech quality than the HVXC, with only 40% complexity of the HVXC.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.10 no.1
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• pp.121-127
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• 2010

This paper relates with the improved CMA blind adaptive equalization algorithm which uses the constellation matching method that improve the inverse modelling efficiency of a communication channel compared to the present CMA blind adaptive equalizer. The amplitude distortion can be compensated in the present CMA blind adaptive equalizer which is used for the reduction of intersymbol interference by distortion that generate such as a band limited wireless mobile channel, but in the improved adaptive alogorithm operates with the minimize the amplitude phase distortion in the output of equalizer by applying the cost function that is composition of additional signal constellation matching error terms. In order to evaluation of the inverse modeling efficiency of improved algorithm, the residual intersymbol interference and recovered signal constellation were compared by computer simulation. As a result of comparion of computer simulation, the improved algorithm has a good stability in the residual intersymbol interference in the steady state, but it has a slow convergence rate in the adaptation state in initial state.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.32 no.3C
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• pp.249-255
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• 2007

In general, a large amount of the memory accesses are required for the CAVLC decoding in H.264/AVC. It is a serious problem for the applications such as a DMB and videophone services because the considerable power is consumed for accessing the memory. In order to solve this problem, we propose an efficient decoding method for the coeff-token which is one of the syntax elements of CAVLC. In this paper, the variable length code table is re-designed with the new codewords which are defined by investigating the architecture of the conventional codeword for the coeff_token element. A new coeff_token decoding method is developed based on the suggested table. The simulation results show that the proposed algorithm achieves an approximately 85% memory access saving without video-quality degradation, compared to the conventional CAVLC decoding.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2011.11a
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• pp.110-112
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• 2011

GPU의 병렬성과 연산능력을 일반적인 공학적 문제 해결에 적용하는 GPGPU 컴퓨팅에 대한 연구가 최근 활발히 진행되고 있다. 비디오 압축과정에는 많은 양의 화소 데이터에 동일하게 반복되는 연산을 수행하는 알고리즘이 많이 적용되므로 GPGPU를 통한 고속 병렬 계산의 응용 분야로 매우 적합하다. H.264/AVC는 비디오를 압축하는 가장 최신의 국제표준으로 여러 제품군과 서비스에 대한 적용되어 시장에서 널리 사용되고 있다. 본 논문에서는 GPGPU의 응용 분야로 주목 받고 있는 비디오 압축 분야에 대한 적용으로 H.264/AVC의 화면내 예측 모드 결정과정에 GPGPU 병렬 프로그래밍을 적용하여 예측 모드 결정 속도를 향상하는 방법을 제안한다. GPU상에서의 데이터 병렬처리를 위해 CUDA C언어를 사용하였으며, CPU상에서의 연산은 C언어를 사용하여 구현되었다. GPU상에서 프레임 전체에 대한 화면내 예측 모드를 병렬적으로 결정함으로써 이에 소요되는 시간을 줄여 줄 수 있었다. 실험결과 GPU상에서 병렬적으로 예측 모드를 결정할 때 Full-HD급 영상에서 약 2.8배 정도의 속도 향상을 확인할 수 있었다. 향후 GPGPU 병렬 프로그래밍을 화면 내 예측뿐만 아니라 반복되는 연산을 수행하는 다른 알고리즘에도 적용하여 부호화기의 계산 부담을 덜어준다면 고속 실시간 비디오 압축 부호기 개발이 더욱 용이해 질것으로 기대된다.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.40 no.6
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• pp.61-70
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• 2003

In this paper, a transcoding algorithm for the Selectable Mode Vocoder (SMV) and the G.723.1 speech coder 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 process. The proposed algorithm is composed of four parts: the parameter decoding, line spectral pair (LSP) conversion, pitch period conversion, excitation conversion and rate selection. The evaluation results show that the proposed algorithm achieves equivalent speech quality to that of tandem transcoding with reduced computational complexity and delay.

• The KIPS Transactions:PartC
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• v.10C no.2
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• pp.233-240
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• 2003

In this paper, an improved architecture of turbo codec for IMT-2000 is proposed. The encoder consists of an interleaver using an on-the-fly type address generator and a modified shift register instead of an external RAM, and the decoder uses a decreased number of RAM. The proposed architecture is simulated with C/VHDL languages, where BER (bit-error-rate) performances are generally in agreement with previous data by varying interaction numbers, interleaver block sizes and code rates.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.169-172
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• 1999

본 논문에서는 EVRC 음성 부호화 시스템을 $OakDSPCore^{\circledR}$를 기반으로 설계된 C&S Technology사의 CSD17C00 칩을 이용하여 전 과정을 어셈블리어로 실시간 구현하였다. 구현된 EVRC 음성 부호화기는 최대의 계산량을 요구하는 8kbps일때 잡음제거 알고리즘을 제외한 인코더부분이 평균 22.5MIPS 이며, 디코더부분은 약 3.35MIPS의 복잡도를 나타낸다. 사용된 메모리양은 프로그램 ROM 10.8K words 데이터 ROM(table) 6.72K words 및 RAM 2.94K words이다. 구현된 EVRC 음성 부호화기는 북미 표준화 기구인 TIA(Telecommunications Industry Association)에서 제공하는 19 개의 test 백터들을 모두 통과하였다.