• Journal of Broadcast Engineering
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• v.20 no.4
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• pp.533-544
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• 2015

Recently, most of the video services are usually transmitted in wireless networks. In networks environment, a packet of video is likely to be lost during transmission. For this reason, this paper proposes a new Error Concealment (EC) algorithm. For High Efficiency Video Coding (HEVC) bitstreams, the proposed algorithm includes Adaptive Prediction Unit-based Motion Vector Extrapolation (APMVE) and Boundary Matching (BM) algorithm, which employs both the temporal and spatial correlation. APMVE adaptively decides a Error Concealment Basic Unit (ECBU) by using the PU information of the previous frame and BM employing the spatial correlation is applied to only unreliable blocks. Simulation results show that the proposed algorithm provides the higher subjective quality by reducing blocking artifacts which appear in other existing algorithms.

• Journal of Broadcast Engineering
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• v.13 no.5
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• pp.719-731
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• 2008

This paper defines QoE (Quality of Experience) for multi-view video streaming service over WiBro and proposes the CLO (Cross-Layer Optimization) algorithm can maximize this. Proposal CLO algorithm contains from physical layer to video layer. Under the time-varying wireless channel condition, the CLO technique takes view-wise and the temporal priority of the multi-view video into consideration in order to decide the transmission of frames and its FEC level. At the handover situation, it is shown through computer simulation that the optimal quality of the multi-view video can be achieved using the minimum amount of resources if the proposed CLO technique is applied.

• Journal of Broadcast Engineering
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• v.13 no.5
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• pp.612-621
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• 2008

Due to the various constraints such as feasible size of data payload and low transmission power, no technical specifications on the voice communication are included in the Zigbee standard. In this paper, a voice coding technique for application to the IEEE 802.15.4 standard, which is the basis of Zigbee communication, is presented. Here, both high compression and good waveform recovery are essential. To meet those requirements, a multi-stage discrete wavelet transform (DWT) block and a binary coding block consisting of two different pulse-code modulations are exploited. Theoretical analysis and simulation results in an indoor wireless channel show that the voice coder with 2-stage DWT is most appropriate from the viewpoint of compression and waveform recovery. When the line-of-sight component is dominant, the voice coding scheme has good recovery capability even in the moderate signal-to-noise power ratios. Hence, it is considered that the presented scheme will be a technical reference for the future recommendation of voice communication exploiting Zigbee.

• Journal of Broadcast Engineering
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• v.19 no.1
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• pp.44-55
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• 2014

The goal of 3D stereoscopy is not only to maximize positive experiences (such as sense of realism) by adding depth information to 2D video but to also minimize negative experiences (such as fatigue). This study examines the impact of different depth levels induced by adjusting 3D camera convergences on positive and negative experiences and finds an optimal parameter for viewers. The results show that there are significant differences among depth levels on spatial involvement, realistic immersion, presence, depth perception, screen transmission, materiality, shape perception, spatial extension and display perception. There are also significant differences for fatigue and unnaturalness. This study suggests that reducing the camera convergence angle of an object by $0.17^{\circ}$ behind the object is the optimal parameter in a 3D stereoscopic setting.

• Journal of Broadcast Engineering
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• v.18 no.5
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• pp.680-690
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• 2013

In this paper, we propose a real-time temporal synchronization and compensation algorithm in stereoscopic video. Many temporal asynchronies are caused in the video editing stage and due to different transmission delays. These temporal asynchronies can degrade the perceived 3D quality. The goal of temporal alignment is to detect and to measure the temporal asynchrony and recover synchronization of the two video streams. In order to recover synchronization of the two video streams, we developed a method to detect asynchronies between the left and the right video streams based on a novel spatiogram information, which is a richer representation, capturing not only the values of the pixels but their spatial relationships as well. The proposed novel spatiogram additionally includes the changes of the spatial color distribution. Furthermore, we propose a block-based method for detection of the pair frame instead of one frame-based method. Various 3D experiments demonstrate the effectiveness of the proposed method.

• Journal of the Korea Society of Computer and Information
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• v.12 no.4
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• pp.137-148
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• 2007

Due to the advance of technology, the technology of various media is rapidly converged into digital technology. However, the conversion into digital radio is not achieved in korea yet. This paper is to suggest the best method of digital conversion of radio in korea by examining some conversion examples in foreign countries. In addition, the survey results on radio audience attitude is analysed by 2,000 audience samples in Korea based on the literature analysis. Radio is the unique media which have the strengths such as mobility, individuality, presence, broad coverage and instant. Most of all, radio cannot help keeping continuously the public broadcasting service in the future. So, the conversion into digital radio is crucial to sustain the competitiveness in digital media environment. This paper suggested that the digital conversion of radio broadcasting needs DRM transmission and should pursuit the policy which is distinguished between a AM and a standard FM, and between domestic broadcast and an international broadcast. Additionally, this paper proposed that we need to develop the multiple device for receiving of signal and the setting of standard technology.

• Journal of Broadcast Engineering
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• v.9 no.4 s.25
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• pp.402-410
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• 2004

Recently, there has been much interest in realistic broadcasting that is a new field following HDTV and 3DTV. In general. the realistic broadcasting is composed of diverse components such as aquisition, authoring, compression, transmission and display, posing many challenging tasks. It is necessary that the types of realistic contents need to be defined prior to the development of realistic broadcasting systems. Based upon them, other components need to be designed and developed. In this paper, we propose some realistic contents suitable to the realistic broadcasting as well as techniques of generating them. Our proposed contents consist of stereoscopic multiview sequences, object-based stereoscopic images, depth map-based image compositing and the composition of stereoscopic real and graphics images. Content generation techniques and their associated software modules are presented with realistic images produced from our experiments. Those contents are produced to deliver stereoscopic perception, immersion and realism to the users as shown in our experimental results.

• Journal of Broadcast Engineering
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• v.10 no.3
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• pp.286-296
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• 2005

This paper presents a stereo-based image compression and transmission system for 3D realistic television. In the proposed system, a disparity map is extracted from an input stereo image pair and the extracted disparity map and one of two input images are transmitted or stored at a local or remote site. However, correspondences can not be determined in occlusion areas. Thus, it is not easy to recover 3D information in such regions. In this paper, a reconstruction image compensation algorithm based on error block concealment and in-loop filtering is proposed to minimize the reconstruction error in generating stereo image pair. The effectiveness of the proposed algorithm is shown in term of objective accuracy of reconstruction image with several real stereo image pairs.

• Journal of Broadcast Engineering
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• v.10 no.3
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• pp.297-312
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• 2005

The multimedia traffic of continuous video and audio data via streaming service accounts for a significant and expanding portion of the Internet traffic. This streaming data delivery is mostly based on RTP with UDP. However, UDP does not support congestion control. For this reason, UDP causes the starvation of congestion controlled TCP traffic which reduces its bandwidth share during overload situation. In this paper, we propose a new TCP-friendly rate control scheme called 'TF-RTP(TCP-Friendly RTP)'. In the congested network state, the TF-RTP exactly estimates the competing TCP's throughput by using the modified parameters. Then, it controls the sending rate of the video streams. Therefore, the TF-RTP adjusts its sending rate to TCP-friendly and fair share with competing TCP traffics. Through the simulation, we prove that the TF-RTP correctly estimates the TCP's throughput and improves the TCP-friendliness and fairness.

• Journal of Broadcast Engineering
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• v.10 no.3
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• pp.313-320
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• 2005

The increasing popularity of multimedia streaming services introduces new challenges in content distribution. Especially, it is important to provide the QoS guarantees as they are increasingly expected to support the multimedia applications. Multimedia streams typically experience the high start-up delay due to the large protocol overhead, the delay, and the loss properties of the wireless networks. The service providers can improve the performance of multimedia streaming by caching the initial segment (prefix) of the popular streams at proxies near the requesting clients. The proxy can initiate transmission to the client while requesting the remainder of the stream from the server. In this paper, we propose the traffic based caching algorithm (TSLRU) to improve the performance of caching proxy. TSLRU classifies the traffic into three types, and improve the performance of caching proxy by reflecting the several elements such as traffic types, recency, frequency, object size when performing the replacement decision. In simulation, TSLRU performs better than the existing schemes in terms of byte hit rate, hit rate, startup latency, and throughput.