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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.

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.

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.

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.

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 the higher coding performance than the previous video coding standards, high efficiency video coding (HEVC) adopts an angular intra prediction method, which requires heavy computational complexity due to the increased intra prediction modes. In this paper, we propose a fast intra prediction mode decision based on the estimation of rate distortion cost using Hadamard transform to reduce the number of intra prediction mode and early termination whether the current coding unit is splitted or not. The experimental results show that the proposed method reduces the computational complexity of intra prediction in HEVC and achieves similar coding performance to that of HEVC test mode 2.1

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 proposes an efficient media synchronization framework for SVC video transport over IP networks. To support synchronization between SVC video and audio signals transported over IP networks, RTP/RTCP protocol suite is usually employed. To provide a framework for media synchronization, we suggest an efficient RTP packetization mode and propose a computationally simple RTCP packet processing method. With the suggested RTP packetization mode, layer synchronization among scalable layers of the SVC video can be effectively achieved. Also by adopting the computationally simple RTCP packet processing, we do not need to process every RTCP SR packet for inter-media synchronization between video and audio signals.

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.