• Journal of the Korea Computer Industry Society
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• v.2 no.7
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• pp.997-1014
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• 2001

In this paper, we propose a method for dynamic selection of unrestricted motion vector(UMV) or default prediction mode(DPM) in H.263 bit stream. For this, we use the error of compensated image and the magnitude of motion vector. In the proposed strategy, the UMV mode is dynamically applied in a frame according to average magnitude of motion vector and error of compensated image. This scheme has improved the quality of image compared to the fixed mode UMV or DPM only. Number of searching points are greatly reduced when comparing to UMV. The Proposed method is more profitable to long video sequences having camera movement locally.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.9 no.6
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• pp.1195-1201
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• 2005

This paper deals with procedures and results for teal time implementation of G.729 speech coder and channel coder including convolution codec, viterbi decoder, and interleaver using a fixed point DSP chip for radio communication systems. We described the method for real-time implementation based on integer simulation results and explained the implemented results by quality performance and required complexity for real-time operation. The required complexity was 24MIPS and 9MIPS in computational load, and 12K words and 4K words in execution code length for speech and channel. The functional evaluation was performed into two steps. The one was bit exact comparison with a fixed point C code, the other was executed by actual speech samples and error test vectors. Unlik other results such as individual implementation, We implemented speech and channel coders on a DSP chip with 160MIPS computation capability and 64 K words memory on chip. This results outweigh the conventional methods in the point of system complexity and implementation cost for radio communication system.

• The Journal of the Acoustical Society of Korea
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• v.24 no.5
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• pp.248-254
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• 2005

In this Paper, the structure of a split bandwidth wideband speech coder and its highband coder for tone qualify elevation are Proposed. The lowband and highband by the split bandwidth method are encoded independently applying the G.729E and MLT (Modulated Lapped Transform) residual model. In the highband structure which is encoded by low bit rate of 4kbps, the MLT residual signals are distinguished to voice and unvoice signal . The voice signals are applied to MLT peak picking method by lowband pitch period. Because transformed MLT residual signals are represented by periodic signal that have periodic peak. The unvoice signals are applied to MLT which linear prediction spectral response is added and do vector quantization. Performance for proposed 15.8kbps wideband speech coder was verified through subjective listening test.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.22 no.4
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• pp.714-725
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• 1997

This paper proposes an image compression method using the wavelet transform and bit-plane coding of wavelet coefficients. The hierarchical application of wavelet transform to an image produces one low resoluation(the subband with lowest frequency) image and several high frequency subbands. In the proposed method, the low resolution image is compressed by a lossless method at 8 bits per each coefficient. However, the high frequency subbands are decomposed into 8 bit planes. With an adptive block coding method, the decomposed bit planes are effectively compressed using localized edge information in each bit plane. In addition, the propsoed method can control bit rates by selectively eliminating lessimportant subbands of low significant bit planes. Experimental results show that the proposed scheme has better performance in the peak signal to noise ratio (PSNR) and compression rate than conventional image coding methods using the wavelet transform and vector quantization.

• Journal of the Korean Institute of Telematics and Electronics
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• v.27 no.3
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• pp.13-20
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• 1990

This paper describes an interframe image coding using motion compensated and classified vector quantizer (MC-CVQ). It is essential to carefully encode blocks with significant pels in motion compensated vector quantizers (MCVQ). In this respect, we propose a new CVQ algorithm which is appropriate to the coding of interframe prediction error after motion compensation. In order to encode an image efficiently at a low bit rate, we partition each block, which is the processing element in MC, into equally sized 4 vectors, and classify vectors into 15 classes according to the position of significant pels. Vectors in each class are then encoded by the vector quantizer with the codebook independently designed for the class. The computer simulation shows that the signal-to-noise ratio and the average bit rate of MC-CVQ are 35-37dB and 0.2-0.25bit/pel, respectively, for the videophone or video conference type image.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.22 no.10
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• pp.2112-2122
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• 1997

This paper presents a new video codec which is based on 2-D triangular mesh-based motion compensation and two step grid point motion estimation. With this approach the efficiency of compression and the quality of reconstructed images are improved. The compensation of motion for each triangular patch is performed by image warping using motion vectors at the grid points. The prediction error coding and the rate control meet MPEG-4 VM 3.0 specification. The experimental results show that the codec system proposed is simple in complexity and moreover, the quality of decoded images is improved.

• Proceedings of the Acoustical Society of Korea Conference
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• spring
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• pp.153-156
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• 2000

본 논문에서는 멀티미디어 서비스들 중에서 음성 또는 오디오 신호를 저속으로 압축할 때 사용되는 G.723.1 부호화기의 line spectral frequency(LSF) 계수 양자화 방식을 고속으로 처리하는 알고리즘을 제안하였다. 제안된 고속탐색 방법은 LSF 계수의 순서성질을 이용하여 코드북의 탐색 범위를 줄임으로써 계산량을 크게 감소시킨다. 제안된 고속탐색 방법을 predictive split VQ(PSVQ) 구조를 갖는 G.723.1 에 적용한 결과 spectral distortion(SD) 성능 감쇄 및 추가적인 메모리 증가 없이 최적 코드벡터를 찾기 위한 코드북 탐색 과정에서 코드북의 평균 탐색 범위가 $20.1\%$ 감소했으며, 이는 additions, subtractions, multiplies 및 comparisons 수가 각각 $19.1\%$, $20.1\%$, $19.4\%$ 및 $12.2\% 감소하는 결과를 얻었다.

• Proceedings of the IEEK Conference
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• 2001.09a
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• pp.861-864
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• 2001

많은 멀티미디어와 DSP 응용에서 입력과 출력 데이터 길이가 같은 고정 길이 곱셈기가 요구된다. 고정 길이 곱셈기는 확률적인 추정에 근거한 적절한 보상 바이어스를 더해줌으로써 일반적인 병렬 곱셈기와 비교하여 50%의 면적을 줄일 수 있다. 본 논문에서는 CSD 곱셈기에 적합한 고정길이 곱셈기의 구조를 제시하고 전파 캐리 선택절차를 이용한 부호확장제거방법과 결합함으로서 새로운 곱셈기구현 방안을 제시한다. 이 곱셈기의 응용으로서 SSB/BPSK-DS/CDMA 전송방식에 사용되는 힐버트 트랜스포머를 43탭 FIR 필터로 구현하고 기존의 compensation 벡터 방법과 비교하여 약 34%의 부호확장 오버헤드를 줄일 수 있음을 보인다.

• Proceedings of the IEEK Conference
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• 2000.09a
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• pp.791-794
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• 2000

본 논문에서는 TMS320C549 를 사용하여 4 채널 G.729A 음성 부호화기를 실시간 구현하였으며, 실제로 음성 사서함 서비스 시스템에 응용하였다. 구현된 G.729A 는 패널 당 부호화기와 복호화기에 각각 14.5MIPS 와 3.6 MIPS 를 소요하였으며, 메모리는 코드와 데이터 부분에 각각 9.88K 워드, 1.69 K 워드를 필요로 하였다. 결과적으로 개발된 VMS 시스템에는 두 개의 DSP 를 사용하여 DSP 당 4 채널씩 총 8 채널을 수용하였다. 실험 결과, ITU-T에서 제공된 모든 테스트 벡터 결과와 비트 단위로 동일하였다.

• Proceedings of the IEEK Conference
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• 2000.09a
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• pp.751-754
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• 2000

본 논문에서는 EVRC 음성 부호화 시스템을 OakDSPCore(R) 를 기반으로 설계된 C&S Technology사의 CSD17C00칩을 이용하여 전 과정을 어셈블리어로 실시간 구현하였다. 구현된 EVRC음성 부호화기는 최대의 계산량이 요구되는8kbps 전송 모드일 때, 인코더부분이 최대24.45MIPS 이며 디코더부분은 3.35MIPS의 복잡도를 나타낸다. 사용된 메모리양은 프로그램 ROM 12.2Kworsd, 데이터 ROM(table) 6.72Kwords 및 RAM2.94Kwords 이다. 구현된 EVRC음성 부호화기는 북미 표준화 기구인Telecommunications Industry Association(TIA)에서 제공하는 19개의 test 벡터들을 모두 통과하였다.