• Journal of Broadcast Engineering
• /
• v.13 no.5
• /
• pp.732-743
• /
• 2008

In H.264/AVC, 4$\times$4 block transform is used for intra and inter prediction instead of 8$\times$8 block transform. Using small block size coding, H.264/AVC obtains high temporal prediction efficiency, however, it has limitation in utilizing spatial redundancy. Motivated on these points, we propose a multi-dimensional transform which achieves both the accuracy of temporal prediction as well as effective use of spatial redundancy. From preliminary experiments, the proposed multi-dimensional transform achieves higher energy compaction than 2-D DCT used in H.264. We designed an integer-based transform and quantization coder for multi-dimensional coder. Moreover, several additional methods for multi-dimensional coder are proposed, which are cube forming, scan order, mode decision and updating parameters. The Context-based Adaptive Variable-Length Coding (CAVLC) used in H.264 was employed for the entropy coder. Simulation results show that the performance of the multi-dimensional codec appears similar to that of H.264 in lower bit rates although the rate-distortion curves of the multi-dimensional DCT measured by entropy and the number of non-zero coefficients show remarkably higher performance than those of H.264/AVC. This implies that more efficient entropy coder optimized to the statistics of multi-dimensional DCT coefficients and rate-distortion operation are needed to take full advantage of the multi-dimensional DCT. There remains many issues and future works about multi-dimensional coder to improve coding efficiency over H.264/AVC.

• Proceedings of the IEEK Conference
• /
• 2000.11d
• /
• pp.105-108
• /
• 2000

본 논문에서 제안한 방법은 DCT 임베이디드 동영상 부호화기를 사용하여 부호화기의 레이트 디스토션 성능과 기존 프레임과 예측 프레임간의 의존성을 이용한 디스토션이 일정한 효율적인 비트율 제어 알고리즘을 제안한다. 다양한 표준 동영상에 대해 컴퓨터 모의 실험을 수행하고 기존 방법과의 비교를 통해 제안방법의 유효성을 검증하고 제안된 알고리즘의 부호화 효율을 확인했다.

• Proceedings of the IEEK Conference
• /
• 2000.09a
• /
• pp.717-720
• /
• 2000

H.263의 시험모델인 TMN5를 최대한 적용하여 실험하였으며 분산, 엔트로피, 움직임 크기 등의 퍼지변수를 데이터 영역에서 추출하여 퍼지화하였다. 소속함수를 계산하기 위해 최소값으로 가장 분명한 퍼지값을 추출하였으며 퍼지집합을 위해서는 각 소속함수로부터의 요소를 더하는 의미에서 최대값을 선택하였다. 무게중심기법을 이용하여 최종 퍼지감도를 구하여 TMN5에 부가하였다.

• Proceedings of the Korean Society of Broadcast Engineers Conference
• /
• 2000.11b
• /
• pp.69-74
• /
• 2000

본 논문에서 제안한 방법은 DCT 임베이디드 동영상 부호화기를 사용하여 부호화기의 레이트 디스토션 성능과 기존 프레임과 예측 프레임간의 의존성을 이용한 디스토션이 일정한 효율적인 비트율 제어 알고리즘을 제안한다. 다양한 표준 동영상에 대해 컴퓨터 모의 실험을 수행하고 기존 방법과의 비교를 통해 제안방법의 유효성을 검증하고 제안된 알고리즘의 부호화 효율을 확인했다.

• Proceedings of the Korea Institute of Convergence Signal Processing
• /
• 2001.06a
• /
• pp.229-232
• /
• 2001

초저속 전송 통신망에서 비디오 전송을 효율적으로 하기 위한 시스템은 낮은 비디오 전송율에서 높은 압축율을 갖는 부호화 방법들이 사용된다. 현재 저속 통신을 위한 부호화 표준으로 대표적인 ITU-T의 H.263은 DCT기반이므로 낮은 비트율에서 블록화 현상이 발생한다. 본 논문에서 제안하는 다중 가중 인자를 이용한 SPIHT 비디오 부호기는 H.263에 비하여 동영상의 PSNR이 우수한 성능을 보였고 기존 SPIHT 비디오 부호기보다도 PSNR이 우수함을 확인하였다.

• Journal of the Institute of Convergence Signal Processing
• /
• v.3 no.3
• /
• pp.14-20
• /
• 2002

In this paper, we proposed the compression technique of the video data using SPIHT(set partitioning in hierarchical trees) based on wavelet and the multi-stream delivery method for best-effort service as fully utilizing the clients bandwidth over the current Internet. The experiment shows that the proposed method provides about 1.5dB better picture quality without block effects than DCT(discrete consine transform) based coding schemes at the same bit rates because of using the wavelet video coder. In addition, this technique implements the multi-stream transmission based on TCP(transmission control protocol). Thus, it is provided with the best-efforts service which is robust to the network jitter problem, and maximally utilizes the bandwidth of the client's.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.28 no.3C
• /
• pp.307-314
• /
• 2003

In the general multiple video object coder, more interested objects such as speaker or moving object is consistently coded with higher priority. Since the priority of each object may not be fixed in the whole sequence and be variable on frame basis, it must be adjusted in a frame. In this paper, we analyze the independent rate control algorithm and global algorithm that the QP value is controled by the static parameters, object importance or priority, target PSNR, weighted distortion. The priority among static parameters is analyzed and adjusted into dynamic parameters according to the visual interests or importance obtained by camera interface. Target PSNR and weighted distortion are proportionally derived by using magnitude, motion, and distortion. We apply those parameters for the weighted distortion control and the priority-based control resulting in the efficient bit-rate distribution. As results of this paper, we achieved that fewer bits are allocated for video objects which has less importance and more bits for those which has higher visual importance. The duration of stability in the visual quality is reduced to less than 15 frames of the coded sequence. In the aspect of PSNR, the proposed scheme shows higher quality of more than 2d13 against the conventional schemes. Thus the coding scheme interfaced to human- eye proves an efficient video coder dealing with the multiple number of video objects.

• Journal of the Korean Institute of Telematics and Electronics T
• /
• v.35T no.2
• /
• pp.35-38
• /
• 1998

We need the packet transmission of voice, data and video according to the development of ATM technology and packet network. In ATM, packet is parted to the constant length data, we called cell, the method is needed to minimize data loss. Because over ATM, we discard error cell without correction in transmission. So, about video, there is layered coding method. We simulated the coding using wavelet transform, which is structurally layed coded, so need not the additional coder.

• Proceedings of the IEEK Conference
• /
• 2002.06d
• /
• pp.29-32
• /
• 2002

In typical block-based video coding, the objective of RC(Rate Control) is to select the quantization parameters so that the encoder produces bits at the rate of the channel and the overall distortion is minimized. To reduce the huge amount of computations required for offline RC, there have been significant efforts to speed up the process of video encoders. Those efforts have been mainly focused on the modes for bit rate and distortion in types of coders, in terms of the quantization parameters. Because previous works related to model based online RC are based on statistics of previous frame, it occurs the problem such that allocates bits unequally without regard to current frame statistics. In this thesis, an equal bit allocation scheme using current frame statistics is proposed.

• Journal of the Institute of Electronics Engineers of Korea SP
• /
• v.45 no.2
• /
• pp.81-89
• /
• 2008

Wideband speech, characterized by a bandwidth of about 7 kHz (50-7000 Hz), provides a substantial quality improvement in terms of naturalness and intelligibility. Although higher data rates are required, it has extended its application to audio and video conferencing, high-quality multimedia communications in mobile links or packet-switched transmissions, and digital AM broadcasting. In this paper, we present a new bandwidth-scalable coder for wideband speech and audio signals. The proposed coder spits 8kHz signal bandwidth into two narrow bands, and different coding schemes are applied to each band. The lower-band signal is coded using the ITU-T G.729/G.729E coder, and the higher-band signal is compressed using a new algorithm based on the gammatone filter bank with an invertible auditory model. Due to the split-band architecture and completely independent coding schemes for each band, the output speech of the decoder can be selected to be a narrowband or wideband according to the channel condition. Subjective tests showed that, for wideband speech and audio signals, the proposed coder at 14.2/18 kbit/s produces superior quality to ITU-T 24 kbit/s G.722.1 with the shorter algorithmic delay.