• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.545-546
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• 2004

서브밴드 분리에 의한 에코확산 오디오 워터마킹법은 호스트 신호를 특정 주파수 대역으로 분리하고, MPEG 심리음향 모델을 이용하여 각 대역별로 삽입되는 워터마크의 파워를 파라미터 설정 함수에 의하여 설정한다. 여기서, 본 방법의 강인성과 비지각성을 좌우하는 것은 파라미터 설정 함수가 된다. 따라서, 본 연구에서는 최대의 강인성과 최소의 음질 열화를 구현하기 위하여 청취실험을 실시하여 최적의 파라미터 설정 함수 설정방법에 대한 검토를 수행하였다.

• Proceedings of the Korea Multimedia Society Conference
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• 2001.11a
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• pp.271-275
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• 2001

본 논문에서는 컴퓨터를 이용한 원격 강의, 원격 학습, 데모 화면 제작 등의 응용들에 필요한 동영상 멀티미디어 시스템의 설계 및 구현을 보인다. 본 논문에서 다루는 연속적으로 변하는 컴퓨터 스크린 이미지는 실세계 비디오와 크기와 영상 특성에 있어 차이점을 가지며 기존의 MPEG 등과 같은 압축 알고리즘이 부적합하다. 시간적으로 변하는 컴퓨터 스크린과 컴퓨터에서 발생하는 오디오로 구성되는 동영상을 저작 재생하는 멀티미디어 시스템 구현 내용과 시스템 성능 평가 결과를 보인다.

• Proceedings of the Korea Multimedia Society Conference
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• 2003.05b
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• pp.69-73
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• 2003

음향압축에 있어서 인간의 청각신경의 특성을 이용하는 방식이 사용되고 있다. 이러한 방법은 심리음향모델(psychoacustical model)에서 도입되었다. 음향압축에서는 이러한 심리음향모델을 사용하여 인간이 지각할 수 없는 한도 내에서 부호화하지 않는 지각음향부호화(perceptual audio coding)사용한다. 지각음향부호화는 분석필터와 합성필터로 각각 부호화 복호화하는데 이것은 필터뱅크(filter bank)로 구현된 서브밴드코더(subband coder) 이다. 본 논문에서는 분석필터와 합성필터에 사용되는 MDCT(Modified Discrete Cosine Transform)와 IMDCT(Inverse Modified Discrete Cosine Transform)를 FPGA에 구현하였다.

• Journal of the Korean Institute of Intelligent Systems
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• v.21 no.5
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• pp.637-644
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• 2011

Being digitalized, the music can be easily purchased and delivered to the users. However, there is still some difficulty to find the music which fits to someone's taste using traditional music information search based on musician, genre, tittle, album title and so on. In order to reduce the difficulty, the contents-based or the emotion-based music retrieval has been proposed and developed. In this paper, we propose new method to determine the importance of MPEG-7 low-level audio descriptors which are multi-dimensional vectors for the emotion-based music retrieval. We measured the mutual similarities of musics which represent a pair of emotions expressed by opposite meaning in terms of each multi-dimensional descriptor. Then rough approximation, and inter- and intra similarity ratio from the similarity relation are used for determining the importance of a descriptor, respectively. The set of weights based on the importance decides the aggregated similarity measure, by which emotion-based music retrieval can be achieved. The proposed method shows better result than previous method in terms of the average number of satisfactory musics in the experiment emotion-based retrieval based on content-based search.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.25 no.8B
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• pp.1460-1468
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• 2000

The goal of this research is the real-time implementation of MPEG-1 Layer III audio decoder using the fixed-point digital signal processor of TMS320C6201 The main job for this work is twofold: one is to convert floating-point operation in the decoder into fixed-point operation while maintaining the high resolution, and the other is to optimize the program to make it run in real-time with memory size as small as possible. We, especially, devote much time to the descaling module in the decoder for conversion of floating-point operation into fixed-point operation with high accuracy. The inverse modified cosine transform(IMDCT) and synthesis polyphase filter bank modules are optimized in order to reduce the amount of computation and memory size. After the optimization process, in this paper, the implemented decoder uses about 26% of maximum computation capacity of TMS320C6201. The program memory, data ROM, data RAM used in the decoder are about 6.77kwords, 3.13 kwords and 9.94 kwords, respectively. Comparing the PCM output of fixed-point computation with that of floating-point computation, we achieve the signal-to-noise ratio of more than 60 dB. A real-time operation is demonstrated on the PC using the sound I/O and host communication functions in the EVM board.

• The Journal of the Acoustical Society of Korea
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• v.16 no.5
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• pp.43-49
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• 1997

Generally, echo cancelers of a TV conference system or a audio conference system are to carry out a real time processing in the case of the closed room having long reverberation time because the system requires much time to modify filter coefficients to environmental changes. Therefore this paper proposes a new subband adaptive filtering method using polyphase filter banks of MPEG(Moving Picture Experts Group) audio system to solve the problems. This method divides signal spectra of input and output into several frequency bands, and each band is adaptively filtered by using ES-NLMS (Exponential Step-Normalized Least Mean Square) algorithm. The optimal number of subband is determined by computational simulations. According to the results of simulation, ERLE of the subband model is 2dB smaller than general full band, calculation rate's of the subband model is decreased about 88%.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.23 no.4
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• pp.821-835
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• 1998

The Digital Audio-Visual Council(DAVIC) that was established to develop useful multimedia communication services has completed the specifications for providing on-demand services such as Movie on Demand(MoD), Teleshopping and accepting Internet service. And then they are proceeding the works to suport converstional communcation services like Plain Old Telecphone Service(POTS), Video telephone, Video teleconferencing. In this paper, we prpose an efficient terminal architecture which can provide conversational multimedia communication services on DAVIC Set-Top Box (STB) environments. To apply the implemented conversational terminal to the multipoint communication environment, we considered the factors of Qurlity of Services(QoS) that determine grade of conversational communication service. We also present the inter-working scheme and that system structure to satisfy QoS by using new MPEG video bridge which gurantees end to end delay requirements as major element of QoS for achieving the real time communication and does not accompany visual quality degradation.

• The Journal of the Acoustical Society of Korea
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• v.28 no.2
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• pp.148-154
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• 2009

In audio coding, SBR technology synthesizes the high-bands using patched time-frequency information from low-bands and the correction parameters, Since SBR transmits only correction parameters for high-bands, it provides a low-rate coding of high-bands, and is used as a core module of MPEG-4 HE-AAC, SBR was originally designed for audio signal and its performance for speech signal tends to decrease, and the major reason is an excessive noise floor in high-bands which is caused by incorrect tonality computation, In this paper, a new method to determine noise floor level in an adaptive fashion according to the speech characteristics is proposed in order to solve the problem of SBR for speech signal, The proposed method maintains the compatibility with the standard SBR, and the subjective performance evaluation shows that the proposed method improves the SBR performance especially for male speech signal compared with the standard SBR.

• Journal of Broadcast Engineering
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• v.15 no.2
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• pp.205-216
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• 2010

In this paper, a method of MPEG-2 Transport Stream (TS) multiplexing for Ultra HDTV (UHDTV) and its design and implementation as a SW tool is described. In practice, UHD video may be divided into several HD videos and each video is encoded in parallel. Therefore, it is necessary to synchronize and multiplex multiple bitstreams encoding each HD video for transmitting and storing UHD video. In this paper, it is assumed that 4 HD videos partitioning a UHD spatially are encoded as H.264/AVC and two 5.0 channel audios are encoded by AC-3. Therefore, 4 H.264/AVC elementary streams (ESs) and 2 AC-3 ESs is mainly considered in the TS multiplexing of UHD. For the carriage of H.264/AVC and AC-3 over MPEG-2 TS, PES packetization and TS multiplexing are designed and implemented based on the extended specification of the MPEG-2 Systems and ATSC (Digital audio compressed standard), respectively. The implemented UHD TS multiplexing tool emulates real time HW operation in the time unit corresponding to the duration of one TS packet transmission in a given TS rate. In particular, in order to satisfy the timing model, the buffers defined in the TS System Target Decoder (T-STD) are monitored and their statuses are considered in the scheduling of TS multiplexing. For UHD multiplexing, two kinds of multiplexing structures, which are UHD re-multiplexing and UHD program multiplexing, are implemented and their strength and weakness are investigated. The developed UHD TS multiplexing tool is tested and verified in terms of the syntax and semantics conformance and functionalities by using a commercial analyzer and real-time presentation tools.

• 한국HCI학회:학술대회논문집
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• 2006.02a
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• pp.413-418
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• 2006

개인용 미디어 단말로서 MP3 플레이어는 이제 음악 파일을 감상하기 위한 일상 생활에서 중요한 휴대 필수품이 되었으며 단순히 음악을 재생하는데 그치지 않고, 각종 정보들을 표시하기 위한 작은 화면이 부착되고 있으며 더 나아가 이미지 정보나 동영상을 표현할 수 있는 기능들이 추가 되고 있다. 이처럼 변화하는 멀티미디어 서비스에 발맞추어 MPEG 에서는 Multimedia Application Format (MAF) (ISO/IEC 23000) 라는 새로운 표준안을 제안하였다. MAF 표준안의 기본 방식은 산업 현장 및 사용자의 요구에 빠르게 부응하기 위해, 기존에 존재하는 표준안들을 결합시켜 새로운 멀티미디어 파일 포맷을 정립하였다. 그 첫 번째 결과물로서, 이미 널리 쓰이고 있는 MP3 파일 포맷에 새로운 기능을 추가하기 위한 Music Player MAF 포맷의 FDIS 가 제안되었다. Music Player MAF 는 MPEG-1 Audio Layer III(MP3) 와 MPEG-7 MDS 메타데이터 정보를 결합한 파일 포맷으로. 기존 ID3 태그에서 표현되는 정보 보다 훨씬 풍부한 메타데이터 정보와 더불어 선택적으로 하나의 JPEG 이미지를 포함한 형태로 이루어져 있다. 그러나, 현재의 파일 포맷은 시간이 고려되지 않은 하나의 JPEG 이미지만을 포함할 수 있기 때문에, 오늘날 사용자가 요구하는 다양한 멀티미디어 서비스를 제공하기에는 많은 한계점을 갖고 있다. 본 논문에서는 Music Player MAF FDIS 에 제안된 세가지 형태의 Music Player MAF 파일 포맷에 관해 기술 한다. 복수의 JPEG 이미지 및 텍스트를 저장하기 위하여 각각의 파일 포맷이 갖는 문제점을 언급하고 하나 이상의 JPEG 이미지와 자막 정보를 MP3 음악 정보와 동기화 시켜 추가 할 수 있는 기능을 추가를 제안한다. 또한 제안된 파일 포맷을 쉽게 생성할 수 있도록 기존의 XMT-O 스키마를 기반으로 MAF 를 위하여 새롭게 MAF XMT 스키마를 정의하고 정의된 스키마를 기반으로 구현된 Music Player MAF 의 저작툴과 제안된 확장 뮤직 플래이어 MAF 을 위한 재생툴을 구현한다.