• The Journal of the Acoustical Society of Korea
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• v.12 no.1E
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• pp.32-38
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• 1993

본 논문에서는 저 전송속도의 음성 부호화기를 제안하였고 컴퓨터 시뮬레이션을 통하여 성능분석과 유연성을 입증하였다. 제안된 부호화 방식은 입력 음성신호의 Stationarity에 따라 전송 프레임의 길이를 가변하고, 전송 프레임의 대표적인 특징 벡터를 Vector Quatization으로 부호화하였다. 제안된 부호화 방식에서 특징 벡터열은 입력 음성신호를 샘플단위로 Prewindowed RLS Lattice 알고리즘을 통해 구한 PARCOR 계수로 구성된다. 입력 음성신호는 Subsegment로 분할되고, 각 Subsegment에서 대표적인 PARCOR 계수를 구한다. Likelihood Ratio Distortion Measure를 사용하여 유사도에 따라 Subsegment를 병합함으로써 전송프레임을 결정한다. 컴퓨터 시뮬레이션 결과로부터 제안된 VTEL 음성 부호화 방식은 좋은 음질을 유지하면서 전체 전송속도를 크게 줄일 수 있다.

• Annual Conference of KIPS
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• 2004.05a
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• pp.1577-1580
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• 2004

여러 응용 서비스를 유 무선을 포함한 다양한 네트웍을 통해 제공하기 위해서는 네트웍에 적응할 수 있는 서비스 형태가 요구된다. 그 가운데 멀티미디어 서비스의 경우 네트웍이 서로 다른 환경에 적응할 수 있는 해결책으로 트랜스코딩 기술이 제시되었다. 하지만, 트랜스코딩을 위해 필요한 복호 부호의 처리 과정은 실시간으로 제공되는 멀티미디어 스트리밍의 경우에 제약조건으로 작용하고, 이에 따른 처리 과정을 대폭 줄이는 일부 기술들은 사용자 측의 서비스 품질에 문제점을 안고 있다. 본 논문에서는 트랜스코딩을 통한 처리 과정과 사용자 측 서비스 품질의 두 가지 측면을 고려하는 frame dropping 시의 모션 벡터 합성 기법에 대해서 언급한다. 또한, 본 논문에서는 기존의 기법과는 달리 양방향 예측 프레임이 포함된 경우에도 적용할 수 있는 확장성을 제공한다.

• Journal of the Institute of Electronics and Information Engineers
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• v.52 no.7
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• pp.107-118
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• 2015

Although most current video coding standards set a fixed motion vector resolution like quarter-pel accuracy, a scheme supporting multiple motion vector resolutions can improve the coding efficiency of video since it can allow to use just required motion vector accuracy depending on the video content and at the same time to generate more accurate motion predictor. However, the selected motion vector resolution for each motion vector is a signaling overhead. This paper proposes a contradiction testing-based signaling scheme of the motion vector resolution. The proposed method selects a best resolution for each motion vector among multiple candidates in such a way to produce the minimum amount of coded bits for the motion vector. The signaling overhead is reduced by contradiction testing that operates under a predefined criterion at both encoder and decoder with a purpose of pruning irrelevant candidate motion vector resolutions from signaling responsibility. Experimental results verified that the proposed scheme is effective in reducing coded motion information by achieving its $Bj{\o}ntegaard$ delta bit rate (BDBR) gain of about 4.01% on average (and up to 15.17%) compared to the conventional scheme with a fixed motion vector resolution.

• Proceedings of the Korean Institute of Communication Sciences Conference
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• 2004.07a
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• pp.343-343
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• 2004

• Proceedings of the Acoustical Society of Korea Conference
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• 1995.06a
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• pp.164-167
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• 1995

RPE 음성부호화기에서 합성 필터로 인한 구동벡터 양자화잡음의 증폭효과를 분석하고 regular pulse 시퀀스의 양자화로 인한 성능감쇄를 줄이기 위해 pyramid vector 양자화방식을 도입하였다. 제안된 방식의 성능평가는 구동시퀀스 양자화를 위해 adaptive PCM을 이용하는 GSM 표준 RPE 방식과의 객관적 및 주관적 성능비교를 통해 수행하였다.T JDSMDQLRY 결과 제안된 방식은 대략 1dB의 SNR 및 segmental SNR 값 증가를 가져왔고, 또한 비공식 청취시험결과 명료도의 증가를 느낄 수 있었다.

• Annual Conference of KIPS
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• 2004.05a
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• pp.757-760
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• 2004

동영상은 포함하고 있는 정보량이 아주 많기 때문에, 영상처리기술 중에서도 데이터 압축기술은 대단히 중요하다. 본 논문에서는 DCT 이 외의 수법을 이용할 때의 동영상의 부호화에 대하여 검토하고, 특이값 전개에 주목한다. 이 방식이 DCT가 갖는 문제점을 모두 해결할 수 있는 것은 아니지만, 다른 방식에서의 영상부호화의 가능성을 보이고자 한다.

• Journal of Korea Multimedia Society
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• v.1 no.2
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• pp.194-203
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• 1998

In this paper, a new DCT-VQ method is proposed which can solve the problems of VQ such as the degradation of edge and enormous calculations. VQ is carried in DCT domain but spatial domain in order to protect the degradation of edge. DCT makes high correlated image data decorrelated and the energy concentrated on a few coefficients. In DCT domain, the DC coefficient is quantized with 8 bits uniform scalar quantizer and the AC coefficients are divided to three regions and coded with vector qiantizer for considering edge components. For the decrease of the calculation and memory, the vectors for three region have small dimension of $1{\times}7$ and use the same codebook. Thus, the proposed method can fully express the edge components by considering AC coefficients in DCT domain and decrease the calculation and memory be reducing the dimension of vectors.

• The Journal of the Acoustical Society of Korea
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• v.21 no.2
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• pp.167-172
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• 2002

The EVRC (Enhanced Variable Rate Codec) is currently in service as a speech cosec in digital cellular systems in North America and Korea. In the EVRC, the LSP (Line Spectral Pairs) related to energy distribution of speech signals in the frequency domain are coded by weighted split vector quantization. Considering that the LSP codebooks might be trained with the language of the develop country of the codebooks or English, it is expected that codebooks trained with Korean provide the performance improvements in the communication in Korean. In this paper, the EVRC LSP codebooks are designed with korean adopting the LBG algorithm based vector quantization, and the performance improvement of the vector quantization and the accompanying speech quality improvement are demonstrated by spectral distortion, SNR and SegSNR measurements, respectively.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.39 no.6
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• pp.696-708
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• 2002

In this paper, a novel transcoding algorithm for the Adaptive Multi Rate(AMR) and the Enhanced Variable Rate Codec(EVRC) 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. The proposed algorithm consists of the parameter decoding, frame classification, mode decision, and transcoders for two frame types. The transcoders convert the parameters such as LSP, frame energy, pitch delay for the adaptive codebook, fixed codebook vector, and codebook gains. Evaluation results show that while exhibiting better computational and delay characteristics, the proposed algorithm produces equivalent speech quality to that produced by the tandem transcoding algorithm.

• The Journal of the Acoustical Society of Korea
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• v.23 no.3
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• pp.262-267
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• 2004

AMR-WB (Adaptive Multi Rate Wideband) speech codec, the most recent voice codec standardized by 3GPP, has the wider audio bandwidth of 50∼7000 Hz and operates on nine speech coding bit rates between 6.60 and 23.85 kbit/s. This Paper presents the real-time implementation of AMR-WB speech codec by using a 16 bit fixed-point TeakLite DSP. The implemented AMR-WB codec requires the complexity of 52.2 MIPS at 23.85 kbit/s mode and also needs the program memory of 17.9 kwords, data RAM of 11.8 kwords, and data ROM of 10.1kwords. It was verified through passing the all test vectors provided by 3GPP with maintaining bit exactness. Stable operations on the real-time testing board were also proved without any distortions and delays for the audio in/out.