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This paper presents a keypoint detection method based on the Laplacian of Gaussian (LoG). In contrast to the Difference of Gaussian (DoG)-based keypoint detection method used in Scale Invariant Feature Transform (SIFT), we focus on the LoG operator and its higher order derivatives. We provide mathematical analogies between higher order DoG (HDoG) and higher order LoG (HLoG) and experimental results to show the effectiveness of the proposed HLoG-based keypoint detection method. The performance of the HLoG is evaluated with four different tests: i) a repeatability test of the keypoints detected across images under various transformations, ii) image retrieval, iii) panorama stitching and iv) 3D reconstruction. The proposed HLoG method provides comparable performance to HDoG and the combination of HLoG and HDoG provides significant improvements in various keypoint-related computer vision problems.

Video stitching technology is widely used to generate Ultra Wide Vision (UWV) and 360VR. However, preserving spatial-temporal coherence and reducing parallax are still challenging problem. In order to increase stitching quality, we propose a parallax-tolerant real-time video stitching method for 12Kⅹ2K UWV. Our experimental results show that the proposed method reduces the parallax efficiently.

High Efficiency Video Coding (HEVC) has brought significant improvement to video compression in terms of coding performance and subjective quality. However, it is computationally more complex than H.264/Advanced Video Coding (AVC) and, hence, adaptation of HEVC to real-time video services has not been promising. In order to reduce the computational cost, we propose a fast mode decision method that is based on the adaptive ordering of modes. Our experimental results show that the proposed method reduces the total encoding time by 47.27%–60.73% and 47.65%–61.31%, with negligible degradation of coding performance, which is a Bjontegaard delta rate increment of 0.35%–1.04% and 0.64%–1.80% under random-access configuration and low-delay configuration, respectively, compared with HEVC test model 12.1 with fast mode off.

고화질 비디오에 대한 시장의 요구가 높아짐에 따라 고화질 비디오를 기존 보다 낮은 데이터 량으로 압축할 수 있는 새로운 비디오 부호화 표준 기술인 HEVC(High Efficiency Video Coding)가 최근에 개발 완료되었다. 즉 HEVC로 압축한 데이터의 양은 기존 비디오 부호화 표준인 AVC/H.264로 압축한 데이터의 양의 동일 화질 대비 약 50%로 보고되고 있다. 압축 성능이 개선된 새로운 부호화 표준에 대한 시장의 관심은 뜨겁지만 시장에 바로 활용되기 위하여서는 응용 서비스에서 요구하는 수준의 처리속도를 만족시켜야 한다. 다수의 코어가 탑재된 컴퓨터 시스템이 널리 보급된 오늘날의 개발 환경에서 부호화 처리 속도를 개선시키기 위해서 여러 각도의 병렬 부호화 적용이 필수이다. 본 논문에서는 HEVC 부호화기에 화면 분할 병렬화와 프레임 수준의 병렬화를 조합하여 적용할 때 코딩 효율 대비 병렬화로 인한 더 높은 속도 향상 결과를 가져올 수 있는 방법을 제안하였다. 즉 시스템 자원과 병렬로 처리할 프레임에 따라 화면을 적응적으로 분할하게 함으로써 코딩 효율 대비 속도 향상을 개선시킬 수 있었다. 한 화면 안에서는 다수의 타일(Tile) 단위로 병렬처리하고, 참조되지 않는 프레임(Frame)들을 병렬로 부호화하도록 본 논문을 통해 구현하였으며 Full-HD 및 4K UHD 영상을 이용하여 제안하는 방법이 코딩 효율 대비 병렬화로 인한 속도 향상이 개선되었음을 보여주었다.

A new-generation video coding standard, named High Efficiency Video Coding (HEVC), has recently been developed by JCT-VC. This new standard provides a significant improvement in picture quality, especially for high-resolution videos. However, one the most important challenges in HEVC is time complexity. A quadtree-based structure is created for the encoding and decoding processes and the rate-distortion (RD) cost is calculated for all possible dimensions of coding units in the quadtree. This provides a high encoding quality, but also causes computational complexity. We focus on a reduction scheme of the computational complexity and propose a new approach that can terminate the quadtree-based structure early, based on the RD costs of the parent and current levels. Our proposed algorithm is compared with HEVC Test Model version 10.0 software and a previously proposed algorithm. Experimental results show that our algorithm provides a significant time reduction for encoding, with only a small loss in video quality.

In this paper, an early termination algorithm for merge process is proposed to reduce the computational complexity in High Efficiency Video Coding (HEVC) encoder. In the HEVC, the same candidate modes from merge candidate list (MCL) are shared to predict a merge or merge SKIP mode. This search process is performed by the number of the obtained candidates for the both of the merge and SKIP modes. This may cause some redundant search operations. To reduce this redundant search operation, we employ the neighboring blocks which have been encoded in prior, to check on the contextual information. In this study, the spatial, temporal and depth neighboring blocks have been considered to compute a correlation information. With this correlation information, an early termination algorithm for merge process is suggested. When all modes of neighboring blocks are SKIP modes, then the merge process performs only SKIP mode. Otherwise, usual merge process of HEVC is performed Through experimental results, the proposed method achieves a time-saving factor of about 21.25% on average with small loss of BD-rate, when comparing to the original HM 10.0 encoder.

A fast intra-prediction method is proposed for High Efficiency Video Coding (HEVC) using a fast intra-mode decision and fast coding unit (CU) size decision. HEVC supports very sophisticated intra modes and a recursive quadtree-based CU structure. To provide a high coding efficiency, the mode and CU size are selected in a ratedistortion optimized manner. This causes a high computational complexity in the encoder, and, for practical applications, the complexity should be significantly reduced. In this paper, among the many predefined modes, the intra-prediction mode is chosen without rate-distortion optimization processes, instead using the difference between the minimum and second minimum of the rate-distortion cost estimation based on the Hadamard transform. The experiment results show that the proposed method achieves a 49.04% reduction in the intra-prediction time and a 32.74% reduction in the total encoding time with a nearly similar coding performance to that of HEVC test model 2.1.

최근 ITU-T/VCEG과 ISO/IEC MPEG은 공동으로 차세대 비디오 압축 표준 기술인 High efficient video coding (HEVC)를 제정하고 있다. HEVC는 H.264/AVC 표준 기술과 비교해 보면 매우 향상된 압축 효율을 보여 주고 있으며, 특히 Full HD 이상의 매우 큰 사이즈의 동영상 및 그에 따른 고화질 비디오 서비스를 주요 목표로 하고 있다. 그러나 품질과 압축 효율 향상을 위해 새로운 구조와 압축 도구들이 추가됨으로써 인코더의 연산 복잡도가 매우 증가되어 있으며, 이에 화질의 열화를 최소한으로 유지하면서 계산량을 감소시키는 연구가 필요해 지고 있는 상황이다. 본 연구에서는 화면 간 예측 시에 부호화 단위의 분할을 효율적으로 조기 종료하는 방법을 제안한다. 제안된 방법은 현재 코딩 단위 (Coding unit)의 움직임 벡터 정보, 율-왜곡 비용, 그리고 상위 코딩 단위에서의 개별 사이즈에 대한 평균 율-왜곡 비용값을 활용하여 분할을 조기에 중단한다. 실험 결과를 통하여 제안된 방법이 기존의 ECU 방법보다 약 10% 정도로 속도 향상이 있으며, HM 참조 소프트웨어와 비교 시 BD-rate 증가는 1.975% 정도로 매우 적게 나타남을 보인다.

&LMN'SVC HTTP 9OP N& QRSTU6VWXQRN YZ [ \ ]1^' (random access) \_` 7. A Y, _`a' bDASH cdNefgsegment index box ('sidx')@haiRAP Synchronization Box ('raps')jk>lm 7. _`g NY ISO n eo\/ 0n N%h3p^'SVC q rQRs0$tgmovie fragmentsN Am[a7.AbstractIn this paper, we propose a scalable random access scheme in SVC based DASH service that enables random access support not only for base layer of SVC but also for enhancement layers. The proposed method includes extension of segment index box (‘sidx’) from DASH standard, as well as new RAP Synchronization Box (‘raps’). Since the proposed scheme provides random access service for movie fragments with SVC encoded video layers, adaptive scalable random access service is possible. Keyword: MPEG DASH, random access, SVC video, HTTP streaming

In this paper, we propose an estimation of network jitter that occurs when transmitting TCP packets containing MPEG-2 TS in progressive streaming service over wired or wireless Internet networks. Based on the estimated network jitter size, we can calculate required de-jitter buffering time to absorb the network jitter at the receiver side. For this purpose, by exploiting the PCR timestamp existing in the TS packet header, we create a new timestamp information that is marked in the optional field of TCP packet header to estimate the network jitter. By using the proposed de-jitter buffering scheme, it is possible to employ the conventional T-STD buffer model without any modification in the progressive streaming service over IP networks. The proposed method can be applicable to the recently developed international standard, MPEG DASH (dynamic adaptive streaming over HTTP) technology.

In this paper, we propose an effective network-adaptive ARQ based error control scheme to provide video streaming services through IP networks where packet error usually occurs. If time delay and feedback channel are allowed, client can request server to retransmit lost packets through IP networks. However, if retransmission is unconditionally requested without considering network condition and number of simultaneous feedback messages, retransmitted packets may not arrive in a timely manner so that decoding may not occur. In the proposed ARQ, a client conditionally requests retransmission based on assumed network condition, and it further determines valid retransmission time so that effective ARQ can be applied. In order to verify the performance of the proposed adaptive ARQ based error control, NIST-Net is used to emulate packet-loss network environment. It is shown by simulations that the proposed scheme provides noticeable error resilience with significantly reduced traffics required for ARQ.

Recently, the standardization of SVC technology which can provide adaptive video quality in diverse service environments has been completed. This paper proposes a layered encryption technique which takes into account the prioritized layer characteristics of SVC and control constant encryption complexity satisfying the target computational complexity. In particular, it analyzes the importance of NAL unit in the SVC video layer and suggests a method to apply appropriate encryption complexity proper for it. The effectiveness of the proposed method is proved through the comparison of time required for encryption and the reconstructed video quality using imperfect decryption key.

Scalable video coding (SVC) has been standardized as an extension of the H.264/AVC standard. This paper proposes a practical real-time transport protocol (RTP) packetization scheme to transport SVC video over IP networks. In combined scalability of SVC, a coded picture of a base or scalable enhancement layer is produced as one or more video layers consisting of network abstraction layer (NAL) units. The SVC NAL unit header contains a (DID, TID, QID) field to identify the association of each SVC NAL unit with its scalable enhancement layer without parsing the payload part of the SVC NAL unit. In this paper, we utilize the (DID, TID, QID) information to derive hierarchical spatio-temporal relationship of the SVC NAL units. Based on the derivation using the (DID, TID, QID) field, we propose a practical RTP packetization scheme for generating single RTP sessions in unicast and multicast transport of SVC video. The experimental results indicate that the proposed packetization scheme can be efficiently applied to transport SVC video over IP networks with little induced delay, jitter, and computational load.

The scalable extension of H.264, known as scalable video coding (SVC) has been the main focus of the Joint Video Team's work and was finalized at the end of 2007. Synchronization between media is an important aspect in the design of a scalable video streaming system. This paper proposes an efficient media synchronization mechanism for SVC video transport over IP networks. To support synchronization between video and audio bitstreams transported over IP networks, a real-time transport protocol/RTP control protocol (RTP/RTCP) suite is usually employed. To provide an efficient mechanism for media synchronization between SVC video and audio, we suggest an efficient RTP packetization mode for inter-layer synchronization within SVC video and propose a computationally efficient RTCP packet processing method for inter-media synchronization. By adopting the computationally simple RTCP packet processing, we do not need to process every RTCP sender report packet for inter-media synchronization. We demonstrate the effectiveness of the proposed mechanism by comparing its performance with that of the conventional method.

In this paper, a test bed of QoS-aware ubiquitous content access (UCA), which allows a user to access and consume multimedia content through convergent networks of broadcasting and communications (WLAN/CDMA) using diverse devices in a seamless and transparent way, is designed and implemented based on the adaptation of scalable and non-scalable content. Especially, the MPEG-21 digital item adaptation (DIA) is employed to find a systematic solution in choosing an optimal adaptation operation under given usage environments in an interoperable way. Furthermore, scalable video coding (SVC) technology is adopted to provide flexible adaptation in spatial, temporal, and quality dimensions in real-time

High Efficiency Video Coding (HEVC) has been developed as the newest video coding standard which achieves the goal of 50% bit-rate reduction with similar video quality compared to the previous standard, H.264/AVC. In order to achieve this improved coding performance, HEVC adopts various advanced techniques, which require heavy encoding complexity. Similar to the previous video coding standards, HEVC has the most computational complexity to perform motion estimation of inter prediction process. Proposed method can select one inter prediction mode and perform the motion estimation only on the selected inter prediction mode. Experimental results show that the prosed method reduces the computational complexity while minimizing the coding loss.

After the success of H.264/AVC, a next generation codec for Ultra-HD resolution video is being developed under the name called High Efficiency Video Coding(HEVC). This paper presents an optimization method for HEVC by performing early termination that preventing further CU split. In our experiments, the early termination method reduces the computational complexity without noticeable coding loss.

In this paper, we propose a new rate control (RC) algorithm for hierarchical video coding of high efficiency video coding (HEVC). To achieve a constant target bitrate transmission, an effective bit allocation method is first introduced, which assigns the number of target bits to each intra-period, group of picture, and frame by considering the hierarchical structure. A quantization parameter (QP) is determined by using the Cauchy distribution based rate-quantization (RQ) model. We further introduce a novel method to estimate the parameters of the RQ model precisely by employing the linear rate model. Experimental results show that the proposed RC algorithm controls the bitrate accurately with negligible degradation of PSNR.

For large resolution video materials used such as in Ultra High Definition TV applications, high efficiency video compression techniques with a low complexity is highly desirable. In this paper, we propose a fast intra prediction method based on the Rate Distortion cost estimation. The proposed method reduce the computational complexity of the new generation encoder with comparable image quality and bit-rate.

This paper presents an efficient audio/video synchronization method that is quite suitable for video telephony. In terms of computational complexity and simplicity of the software structure, the proposed algorithm shows improved results compared to the conventional synchronization algorithm. The derived audio/video synchronization algorithm is compactly expressed in a single decision rule. After simulations with real video telephony system based on Texas Instruments OMAP processor and Nucleus PLUS RTOS, we found the devised algorithm is quite suitable for consumer cellular phones with limited computational resources.