• Proceedings of the Korean Society of Broadcast Engineers Conference
• /
• 2008.11a
• /
• pp.239-242
• /
• 2008

지상파 DMB는 이동 환경에서 QVGA 급의 영상과 스테레오 오디오를 제공하는 방송 서비스로서 2005년 12월부터 본격적으로 서비스되고 있는데, 최근에는 DMB 환경에서 고품질의 영상과 오디오를 제공하려는 기술에 대한 연구가 이루어지고 있다. 지상파 DMB 환경에서 고품질의 영상 또는 오디오를 제공하기 위해서는 기존의 DMB 서비스에 추가적인 데이터들을 전송하는 것이 필요한데, 하나의 지상파 DMB 방송 채널에 할당되는 전송 비트율이 높지 않다는 점을 감안하면, 이러한 추가적인 데이터들을 효율적으로 전송하는 것이 서비스의 상용화 입장에서는 중요한 요소가 될 수 있다. 본 논문에서는 지상파 DMB 환경에서 멀티채널 오디오 서비스를 제공하고자 할 때, 추가적으로 전송되어야 하는 부가정보 스트림의 효율적인 전송을 위한 패킷화 방법을 제안한다. 지상파 DMB 환경에서 멀티채널 오디오 서비스를 제공하기 위한 부가정보 스트림은 일반 오디오 스트림과 마찬가지로 프레임 단위로 생성이 되며, 약 12kbps의 비트율을 가진다. 그러나, 부가정보 스트림을 지상파 DMB 환경에서 전송하기 위하여, MPEG-2 TS로 패킷화하여 전송하게 되면, 부가정보 스트림의 비트율보다 훨씬 높은 약 32kbps의 전송율을 가지게 된다. 본 연구에서는 이와 같은 문제점을 해결하기 위하여, 멀티채널 오디오 서비스를 위해 필요한 부가정보 스트림의 비트율을 분석하고, 이를 바탕으로 하나의 TS 패킷에 하나 이상의 부가정보 프레임을 포함하여 전송하는 방법을 제안한다. 제안한 방법의 성능 검증을 위해 제안한 방법에 따라 하나 이상의 부가정보 프레임을 하나의 TS 패킷에 포함하여 패킷화하는 것을 시뮬레이션하고, 그 결과를 제시하였다.

• The Journal of The Korea Institute of Intelligent Transport Systems
• /
• v.6 no.2
• /
• pp.95-106
• /
• 2007

This paper proposes an effective tempo feature extraction method for music information retrieval. The tempo information is modeled by the narrow-band temporal modulation components, which are decomposed into a modulation spectrum via joint frequency analysis. In implementation, the tempo feature is directly extracted from the modified discrete cosine transform coefficients, which is the output of partial MP3(MPEG 1 Layer 3) decoder. Then, different features are extracted from the amplitudes of modulation spectrum and applied to different music information retrieval tasks. The logarithmic scale modulation frequency coefficients are employed in automatic music emotion classification and music genre classification. The classification precision in both systems is improved significantly. The bit vectors derived from adaptive modulation spectrum is used in audio fingerprinting task That is proved to be able to achieve high robustness in this application. The experimental results in these tasks validate the effectiveness of the proposed tempo feature.

• Journal of Broadcast Engineering
• /
• v.18 no.1
• /
• pp.43-58
• /
• 2013

Richmedia Service means that interactive media service can provide view with various multimedia elements(such as Video, Audio, Text) at same time. Various Multimedia elements can be serviced by Scene Description technology standards like BIFS(Binary Format for Scenes) and LASeR(Light Application Scene Representation). By providing Scene Component information, richmedia service is available to various multimedia services. so users is available to personalized services fitting temporal and spatial options. In conventional technology, when the scene is changed by user or service, mobile deletes the scene of configuration information and makes new scene of configuration information. this is a very inefficient way. In this paper, Propoesed that by using DOM(Document Object Model) method, to pass only the dynamic configuration part, changes scene method.

• Journal of Korea Multimedia Society
• /
• v.17 no.1
• /
• pp.104-112
• /
• 2014

With recent development of virtual reality technology, coupled with the growth of the marine industry, virtual underwater simulation systems are under development in various studies, for educational purposes and to simulate virtual reality experiences. Current literature indicates many underwater simulation systems to date have focused on the quality of visual stimulus delivered through three-dimensional graphics user interface, limiting the reality of the experience. In order to improve the quality of the reality delivered by such virtual simulations, it is crucial to develop multi-sensory technology rather than focus on the conventional audio-visual interaction, which limits experiencer from the sense of underwater immersion and existence within the simulation. This work proposes the immersive multi-sensory effector system, delivering the users with a more realistic underwater experience. The sense of reality perceived was evaluated, as the main factor of the virtual reality system.

• Journal of KIISE:Computer Systems and Theory
• /
• v.26 no.8
• /
• pp.939-948
• /
• 1999

주문형 비디오나 원격 회의 그리고 원격 강의와 같은 멀티미디어 애플리케이션들은 비디오나 오디오와 같은 멀티미디어 트래픽을 유발한다. 이러한 멀티미디어 트래픽의 가장 큰 특성은 폭주성이다. 폭주성은 통신망의 효율을 떨어뜨리는 매우 중요한 요인이며, 따라서 폭주성에 대처할 수 있는 효율적인 대역폭 할당 정책이 통신망 관리에서 매우 중요하다. 본 논문에서는 멀티캐스팅 방식의 온라인 비디오 애플리케이션에 대해서 대역폭 완화 작업이라고 불리우는 효율적이고 유용한 대역폭 할당 방안을 설계하였다. 본 논문에서 제시하는 새로운 대역폭 완화 기법은 비디오 서버에서 이미 전송된 데이타의 특성을 이용하여 대역폭 완화창의 크기를 조절하는 동적인 기법이다. 몇 가지 MPEG 트레이스들로 실험했을 때 기존의 온라인 대역폭 완화 기법과 비슷한 성능을 보이고, 특히 멀티캐스팅 방식의 온라인 비디오 서비스에 대해서는 서버의 자원 효율성을 향상시킴을 알 수 있었다. Abstract All of the multimedia applications such as VOD, teleconferencing, and tele-lecturing invoke multimedia traffic like video or audio traffic. The most important characteristic of these multimedia traffic is the burstiness property. So, bandwidth management is becoming the major part of network management. In this paper, we propose a new and efficient bandwidth management technique called bandwidth smoothing for the multicasting on-line video applications. Our bandwidth smoothing technique reduces as much of the network bandwidth required to transmit on-line video traffic as previously proposed methods, and improves the server's resource utilization especially for the multicast on-line video services.

• Journal of the Institute of Electronics and Information Engineers
• /
• v.51 no.6
• /
• pp.221-227
• /
• 2014

In this paper, we propose a synthesis method that can effectively suppress the ambience which affects tonal components in the PS decoder. Ambience component was obtained by using decorrelation filter and the weighting of the ambience in the decoder was determined through IC parameter. However, since the parameters are extracted in the sub-band domain, a low IC value could be analyzed even if the tonal component is dominant. The quality of the output signal may be degraded. To prevent this problem, the tonality was measured in the downmixed signal and the weighting of the ambience components were adjusted appropriately according to the measured tonality index. The performance of the proposed method was evaluated by simulations. Furthermore, the subjective test was performed and the results confirmed that the proposed method offers improved quality.

• The Journal of the Korea Contents Association
• /
• v.7 no.6
• /
• pp.69-78
• /
• 2007

In recent years, embedded mobile systems have been expanding their application domains from embedded portable devices which only execute a specialized application such as MP3 player or digital camcoder to digital convergence devices which execute more complicated applications converged various functionalities such as video and audio play, digital dictionary, DMB, games, phone, etc. As it requires the increasing hardware performance such as more faster CPU and more larger RAM, display, disk size, it has brought about a corresponding increase in power consumption. However, coupled with relatively small gains in battery capacity over recent years, the importance of software architecture including intelligent power management has become paramount. In this paper, we have ported UbiFOSTM with energy saving techniques on the ARM9-based MBA2440 platform. For energy savings, we adapted the dynamic power management and the device power management schemes based on monitoring workload. Experimental results with some well-known applications show that proposed low power technique could save energy up to 24 %.

• Journal of the Institute of Electronics Engineers of Korea SP
• /
• v.39 no.5
• /
• pp.477-485
• /
• 2002

In this paper, we present an efficient hardware architecture for the H.263 video codec and its VLSI implementation. This architecture is based on the unified interface by which internal hardware engines and an internal RISC processor are connected one another. The unified interface enables the modular design of internal blocks, efficient hardware/software partitioning, and pipelined paralled operations. The developed VLSI supports the H.263 version 2 profile 3 @ level 10, and moreover, both the control protocol H.245 and the multiplexing protocol H.223. Therefore, it can be used for the complete ITU-T H.324 or 3GPP 3G 324M multimedia processor with the help of an external audio codec. Simultaneous encoding and decoding of QCIF format images at a rate greater than 15 frames per second is achieved at 40 MHz clock frequency.

• Journal of Digital Convergence
• /
• v.12 no.3
• /
• pp.269-275
• /
• 2014

The DAB system was designed to provide CD quality audio and data services for fixed, portable and mobile applications with the required BER below $10^{-4}$. However for the T-DMB system with the video service of MPEG-4 stream, BER should go down $10^{-8}$ by adding FEC blocks which consist of the Reed-Solomon (RS) encoder/decoder and convolutional interleaver/deinterleaver. In this paper we propose two types of turbo coded T-DMB system without altering the puncturing procedure and puncturing vectors defined in the standard T-DMB system for compatibility. One(Type 1) can replace the existing RS code, convolutional interleaver and RCPC code by a turbo code and the other one (Type 2) can substitute the existing RCPC code by a turbo code. Simulation results show that two new turbo coded systems are able to yield considerable performance gain after just 2 iterations. Type 2 system is better than type 1 but the amount of performance improvement is small.

• KSII Transactions on Internet and Information Systems (TIIS)
• /
• v.11 no.8
• /
• pp.4077-4091
• /
• 2017

This paper presents the design of a broadcasting scenario and system for an 8K-resolution content. Due to an 8K content is four times larger than the 4K content in terms of size, many technologies such as content acquisition, video coding, and transmission are required to deal with it. Therefore, high-quality video and audio for 8K (ultra-high definition television) service is not possible to be transmitted only using the current terrestrial broadcasting system. The proposed broadcasting system divides the 8K content into four 4K contents by area, and each area is hierarchically encoded by Scalable High-efficiency Video Coding (SHVC) into three layers: L0, L1, and L2. Every part of the 8K video content divided into areas and hierarchy is independently treated. These parts are transmitted over heterogeneous networks such as digital broadcasting and broadband networks after going through several processes of generating signal messages, encapsulation, and packetization based on MPEG media transport. We propose three methods of generating streams at the sending entity to merge the divided streams into the original content at the receiving entity. First, we design the composition information, which defines the presentation structure for displays. Second, a descriptor for content synchronization is included in the signal message. Finally, we define the rules for generating "packet_id" among the packet header fields and design the transmission scheduler to acquire the divided streams quickly. We implement the 8K broadcasting system by adapting the proposed methods and show that the 8K-resolution contents are stably received and serviced with a low delay.