• Journal of Broadcast Engineering
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• v.24 no.2
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• pp.329-340
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• 2019

Recently, broadcasting technologies have evolved as high-quality AV services such as domestic terrestrial UHD(Ultra-High Definition) broadcasting have been increasing, and broadcasting standards have been newly defined. Also, as network technology develops, contents are consumed not only in the country but also the world. Accordingly, content insertion technology, which is a method of providing suitable contents in accordance with the national and local environments, will be needed. This paper proposes a content insertion service system model and synchronization scheme using ATSC(Advanced Television Systems Committee) 3.0 Event Signaling standard under heterogeneous network environment of broadcasting network and internet network based on transmission standard DASH(Dynamic Adaptive Streaming over HTTP)/ROUTE(Real time Object delivery Over Unidirectional Transport) and MMT(MPEG Media Transport) of terrestrial UHD broadcasting. It also verifies that the service operates in an environment that meets the broadcast standard.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2019.06a
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• pp.305-308
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• 2019

최근 네트워크 기술이 발달함에 따라 실시간 스트리밍 서비스의 사용이 증가하고 있다. 이에 따라, 스트리밍에 있어 복잡성 및 대기시간 등의 문제점을 해결하고자 공통 미디어 포맷에 관한 연구가 진행되고 있다. 또한, 통신망을 활용한 고품질 AV(Audio-Video) 콘텐츠 전송이 가능해짐에 따라, 국내 UHD(Ultra High Definition) 방송 전송 표준은 방송망과 통신망을 활용한 하이브리드 서비스 규격을 제정하였다. 하이브리드 서비스는 통신망을 활용함에 따라, 방송망 관련 부가 데이터 및 사용자 맞춤형 콘텐츠 제공과 같은 다양한 서비스가 가능하게 되었다. 본 논문에서는 국내 지상파 UHD 전송 표준인 MMT(MPEG Media Transport)와 모바일 망에 특화된 기능을 제공하기 위해 제안된 Mobile MMT의 시그널링 메시지를 활용한 콘텐츠 삽입 기술을 제안한다. 또한, 이종망 환경에서 콘텐츠 삽입 기술 시스템 모델 및 서비스 시나리오를 제안하고, 제안한 시스템 모델 기반의 검증플랫폼을 통해 제안 기술의 타당성을 확인한다.

• Journal of Broadcast Engineering
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• v.20 no.1
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• pp.68-81
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• 2015

Owing to the latest development of component and media processing technologies, UHDTV as a successor of the HDTV is expected that this will be coming soon realization. Accordingly, an audio technology that provides a 5.1-channel surround sound in home should be contemplating on what services should be provided with the advent of UHDTV era. In fact, however, the market of 5.1-channel audio is struggling, due to the difficulty of installation and maintenance of the multi speakers in a home. Meanwhile, the movie sound market for a long time been used in 5.1 and 7.1-channel sound formats, have changed as Dolby ATMOS, IOSONO, AURO3D etc. are launched one after another with the introduction of hybrid audio technologies that include the ceiling and object-based sounds. This very object-based audio technology is assured to be introduced in the home theater and broadcast audio market, and this change in audio technology is expected to be a breath of pioneering technological advances and market growth from the channel-based audio market that lacks flexibility. In this paper, we will investigate a suitable realistic audio solution for UHDTV, and introduce hybrid audio technologies, which is expected to be an audio technology for UHDTV, and we will describe the hybrid audio content format and reproduction methods in a home and consider the future prospects of realistic audio.

• Journal of Broadcast Engineering
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• v.22 no.6
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• pp.744-754
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• 2017

This paper compares transport overhead of MPEG-2 TS, MMT and ROUTE in order to compare transport efficiency between the DTV and UHDTV. The MPEG-2 TS standard, widely used, was established for multiplexing and synchronizing encoded audio and video, additional information. In recent years, MMT and ROUTE was established as a next generation multimedia transport standard for the new broadcasting communication environment. In this paper, we compare and analyze transport overhead about three protocol. In order to analysis, we captured the UHD A/V stream in real-time broadcasting service using ROUTE and MMT, and we calculated and analyzed transport overhead using the overhead analysis program which was developed in our laboratory. Furthermore, for comparison under the same conditions, we assumed the MPEG-2 TS stream by extracting ES of UHD A/V stream based on the DTV standard. In this paper, we show the results of protocol transport efficiency in case of basic A/V stream except for additional services. And result show that MMT and ROUTE have similar overhead and MPEG-2 TS is relatively small overhead. However, since MPEG-2 TS result does not consider null packets, it is expected that the relative overhead difference will be reduced.

• Journal of Broadcast Engineering
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• v.25 no.4
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• pp.560-568
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• 2020

In recent years, as network technology develops, the usage of streaming services by users is increasing. However, the complexity of streaming services is also increasing due to various terminal environments. Even when streaming the same content, it is necessary to re-encode the content according to the type of service. In order to solve the complexity and latency of the streaming service, Moving Picture Experts Group (MPEG) has standardized the Common Media Application Format (CMAF). In addition, as content transmission using a communication network becomes possible, the Republic of Korea's Ultra High Definition (UHD) broadcasting standard has been enacted as a hybrid standard using a broadcasting network and a communication network. The hybrid service enables various services such as transmitting additional information of contents or providing user-customized contents through a communication network. The Republic of Korea's UHD transmission standard utilizes MPEG Media Transport (MMT), and Mobile MMT is an extension of MMT to provide mobile network-specific functions. This paper proposes a method of inserting CMAF contents suitable for various streaming services using signaling messages of MMT and Mobile MMT. In addition, this paper proposes a model for content insertion system in heterogeneous network environment using broadcasting and communication networks, and verifies the validity of the proposed technology by checking the result of content insertion.

• Journal of Intelligence and Information Systems
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• v.24 no.2
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• pp.21-35
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• 2018

Since the introduction of MP3 players, CD recordings have gradually been vanishing, and the music consuming environment of music users is shifting to mobile devices. The introduction of smart devices has increased the utilization of music through music playback, mass storage, and search functions that are integrated into smartphones and tablets. At the time of initial MP3 player supply, the bitrate of the compressed music contents generally was 128 Kbps. However, as increasing of the demand for high quality music, sound quality of 384 Kbps appeared. Recently, music content of FLAC (Free License Audio Codec) format using lossless compression method is becoming popular. The download service of many music sites in Korea has classified by unlimited download with technical protection and limited download without technical protection. Digital Rights Management (DRM) technology is used as a technical protection measure for unlimited download, but it can only be used with authenticated devices that have DRM installed. Even if music purchased by the user, it cannot be used by other devices. On the contrary, in the case of music that is limited in quantity but not technically protected, there is no way to enforce anyone who distributes it, and in the case of high quality music such as FLAC, the loss is greater. In this paper, the author proposes an audio watermarking technology for copyright protection of high quality stereo music. Two kinds of information, "Copyright" and "Copy_free", are generated by using the turbo code. The two watermarks are composed of 9 bytes (72 bits). If turbo code is applied for error correction, the amount of information to be inserted as 222 bits increases. The 222-bit watermark was expanded to 1024 bits to be robust against additional errors and finally used as a watermark to insert into stereo music. Turbo code is a way to recover raw data if the damaged amount is less than 15% even if part of the code is damaged due to attack of watermarked content. It can be extended to 1024 bits or it can find 222 bits from some damaged contents by increasing the probability, the watermark itself has made it more resistant to attack. The proposed algorithm uses quantization in DCT so that watermark can be detected efficiently and SNR can be improved when stereo music is converted into mono. As a result, on average SNR exceeded 40dB, resulting in sound quality improvements of over 10dB over traditional quantization methods. This is a very significant result because it means relatively 10 times improvement in sound quality. In addition, the sample length required for extracting the watermark can be extracted sufficiently if the length is shorter than 1 second, and the watermark can be completely extracted from music samples of less than one second in all of the MP3 compression having a bit rate of 128 Kbps. The conventional quantization method can extract the watermark with a length of only 1/10 compared to the case where the sampling of the 10-second length largely fails to extract the watermark. In this study, since the length of the watermark embedded into music is 72 bits, it provides sufficient capacity to embed necessary information for music. It is enough bits to identify the music distributed all over the world. 272 can identify $4*10^{21}$, so it can be used as an identifier and it can be used for copyright protection of high quality music service. The proposed algorithm can be used not only for high quality audio but also for development of watermarking algorithm in multimedia such as UHD (Ultra High Definition) TV and high-resolution image. In addition, with the development of digital devices, users are demanding high quality music in the music industry, and artificial intelligence assistant is coming along with high quality music and streaming service. The results of this study can be used to protect the rights of copyright holders in these industries.

• The Journal of the Acoustical Society of Korea
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• v.30 no.4
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• pp.197-206
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• 2011

In this paper, we proposes the virtual sound source panning algorithm in the multichannel system. Recently, High-definition (HD) and Ultrahigh-definition (UHD) video formats are accepted for the multimedia applications and they provide the high-quality resolution pixels and the wider view angle. The audio format also needs to generate the wider sound field and more immersive sound effects. However, the conventional stereo system cannot satisfy the desired sound quality in the latest multimedia system. Therefore, the various multichannel systems that can make more improved sound field generation are proposed. In the mutichannel system, the conventional panning algorithms have acoustic problems about directivity and timbre of the virtual sound source. To solve these problems in the arbitrary positioned multichannel loudspeaker system, we proposed the virtual sound source panning algorithm using multiple vectors base nonnegative amplitude panning gains. The proposed algorithm can be easily controlled by the gain control function to generate an accurate localization of the virtual sound source and also it is available for the both symmetric and asymmetric loudspeakers format. Its performance of sound localization is evaluated by subjective tests comparing with conventional amplitude panning algorithms, e.g. VBAP and MDAP, in the symmetric and asymmetric formats.