• 방송공학회논문지
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• 제6권1호
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• pp.21-31
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• 2001

PN 시퀸스를 삽입하는 오디오 워터마킹은 들리지 않으면서도 강인한 워터마크를 만들기 위해 심리음향모델을 사용하여 PN 시퀸스를 변형시킨다. 하지만 워터마크를 삽입하는 모든 프레임에 대해 심리음향모델을 계산하기 위해서는 부호화 과정이 매우 복잡해지는 문제가 있다 부호화기에서 심리음향모델의 역할을 대신 하도록 만든 고정필터로 PN 시퀸스를 변형하면 훨씬 간단하 면서도 효율적인 워터마킹 시스템이 가능해진다. 본 논문에서는 고정된 지각필터를 도입한 효율적인 워터마킹 시스템을 제안한다. 심리음향모델을 대체하는 고정필터를 사용함으로써 PN 시퀸스를 들리지 않게 만들어주고 따라서 강인한 워터마크를 삽입할 수가 있다. 이와 함께 복호화기에서는 PN 시퀸스를 매칭시켜 상관도를 높여주는 보상필터 구조를 제안하여 원신호를 사용하지 않는 복호화기의 복호화 성능을 향상시켰다.

• 한국방송∙미디어공학회:학술대회논문집
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• 한국방송공학회 2006년도 학술대회
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• pp.289-295
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• 2006

The Musical Slide Show MAF, which currently being standardized by MPEG, conveys the concept of combining several established standard technologies in a single file format. It defines the format of packing up MP3 audio data, along with MPEG-7 Simple Metadata Profile and MPEG-21 Digital Item Declaration metadata; with JPEC images and optional text, and synchronizes them all together to create a slideshow of JPEC image data associated to MP3 audio data during the audio playback. The implementation of Musical Slide Show MAF can be a music karaoke file where users can sing along while listening to the music, view the JPEG slideshow and reading the lyrics; or a story-telling file where users can listen to the narrated story by looking at the related illustration slideshow of the story In this paper we present the tool to producing the Musical Slide Show MAF contents. Regardless the knowledge of user on the MAF file format, the authoring tool simplify the manner of packaging several multimedia contents into single file.

• ETRI Journal
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• 제33권6호
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• pp.897-903
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• 2011

A hybrid ${\Delta}{\Sigma}$ modulator for audio applications is presented in this paper. The pulse generator for digital-to-analog converter alleviates the requirement of the external clock jitter and calibrates the coefficient variation due to a process shift and temperature changes. The input resistor network in the first integrator offers a gain control function in a dB-linear fashion. Also, careful chopper stabilization implementation using return-to-zero scheme in the first continuous-time integrator minimizes both the influence of flicker noise and inflow noise due to chopping. The chip is implemented in a 0.13 ${\mu}m$ CMOS technology (I/O devices) and occupies an active area of 0.37 $mm^2$. The ${\Delta}{\Sigma}$ modulator achieves a dynamic range (A-weighted) of 97.8 dB and a peak signal-to-noise-plus-distortion ratio of 90.0 dB over an audio bandwidth of 20 kHz with a 4.4 mW power consumption from 3.3 V. Also, the gain of the modulator is controlled from -9.5 dB to 8.5 dB, and the performance of the modulator is maintained up to 5 nsRMS external clock jitter.

• 한국음향학회지
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• 제26권5호
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• pp.214-219
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• 2007

본 논문은 MPEG-2 AAC(Advanced Audio Boding) 디코더에 필요한 IMDCT(Inverse Modified Discrete Cosine Transform)를 고속으로 처리하기 위한 새로운 IFFT(Inverse Fast Fourier Transform) 구현 방식을 제안한다. 기존 방식 중에서 $2^n$(N-point) type IMDCT가 성능이 가장 우수하지만 많은 계산을 요구하는 N/4-point complex IFFT 과정을 포함하고 있다. 본 연구는 $2^n$(N-point) type IMDCT에 포함된 N/4-point complex IFFT의 연산량을 줄이는 방법을 고안하였다. N/4-point complex IFFT는 입력 데이터를 bit-reverse 방식을 사용하여 정렬하지만 본 연구에서는 새로운 입력 데이터 정렬방식과 $N/4^{n+1}$ 형태의 IFFT 고안하여 곱셈, 덧셈, ROM 용량을 줄였다.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2000년도 제13회 신호처리 합동 학술대회 논문집
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• pp.229-232
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• 2000

Orthogonal frequency division multiplexing(OFDM) has meanwhile become part of several telecommunicati ons standards, such as satellite and terrestrial digital audio broadcasting(DAB), digital terrestrial TV broad casting(DVB), asymmetric digital subscriber line(ADSL) for high-bit-rate digital subscriber services on twisted-pair channels, and broadband indoor wireless systems. In his paper, we show that OFDM signals contain sufficient structure to accomplish blind channel estimation using second order statistics only. This method doesn't require redundancy as cp in transmitter. And the result is compared with PSAM channel estimation as least square, linear minimum mean square, singular value decomposition.

• 전기학회논문지
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• 제56권5호
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• pp.987-989
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• 2007

This paper describes a digital convergence platform(DCP) which is implemented based on the MPEG-21 multimedia framework. DCP is a newly proposed solution in this research for the convergence service of future home multimedia environment. DCP is a common platform designed to have the feature of reconfigurability, by means of S/W. which supports to run diverse digital multimedia services. A distributed peer to peer service and transaction model is also a new feature realized in DCP using the MPEG-21 multimedia framework. A prototype DCP is implemented to verify its functions of multimedia service and transactions. The developed DCPs are networked with IP clustering storage systems for the distributed service of multimedia. Successful streaming services of the MPEG-2/4 video and audio are verified with the implemented test-bed system of DCP.

• 한국멀티미디어학회논문지
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• 제9권12호
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• pp.1617-1627
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• 2006

The concept of middleware for digital TV receivers is not new one. Using middleware for digital TV development has a number of advantages. It makes it easier for manufacturers to hide differences in the underlying hardware. It also offers a standard platform for application developers. Digital TV middleware enables set-top boxes(STBs) to run video, audio, and applications. The main concern of digital TV middleware is now to reduce its memory usage because most STBs in the market are small footprint. In this paper, we propose several ideas about how to reduce the required memory size on the runtime area of DTV middleware using a new native process technology. Our proposed system has two components; the Efficient Native Process Module, and Enhanced Native Interface APIs for concurrent native modules. With our approach, the required memory reduced from 50% up to 75% compared with the traditional approach. It can be suitable for low end STBs of very low hardware limitation.

• 한국전자파학회지:전자파기술
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• 제11권3호
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• pp.45-55
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• 2000

DAB(Digital Audio Broadcasting)는 기존 아날로 그 방식 AM과 FM 라디오 방송의 한계와 문제점을 극복하고 CD 수준 음질의 오디오를 포함한 다양한 멀티미디어 서비스를 제공하는 차세대 라디오 방송이다. DAB는 전송 주파수 대역에 따라 기존의 FM/AM 대역과 다른 새로운 대역을 사용하는 Out-of-Band 방식과 기존의 FM/AM 대역을 사용하는 In-Band 방식으로 분류할 수 있다. Out-of-Band 방식으로는 유럽의 Eureka-147과 일본의 협대역 ISDB-T(Integrated Services Digital Broadcasting-Terrestrial) 방식이 있으며, In-Band 방식은 미국의 USADR(USA Digital Radio)사와 LDR(Lucent Digital Radio)사가 개발하고 있는 IBOC(In-Band On Channel) 방식과 국내 한국전자통신연구원(ETRI)에서 개발 중인 IBAC(In-Band Adjacent Channel) 방식이 있다. 본 고에서는 DAB의 전송 주파수 대역에 따른 방식별 특성을 고찰하고 유럽, 일본, 미국과 국내에서 상용화 서비스 또는 개발되고 있는 DAB 시스템의 기술동향을 분석 한다.

• 한국정보통신학회논문지
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• 제8권7호
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• pp.1488-1492
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• 2004

The digital signal has been replaced the analog signal in most of every field of multimedia including still image, animation, and audio due to the enormous extension of computer supply and the fast development of computer network. The consumers of information are able to enjoy the abundance of information because of one of the digital signal traits that very easy to regenerate the original data. Because of the trait, however, it is very hard for the producers of information to keep the copyright with the merit of original copy in quality excellency. In this paper, the watermarking technology which inserts a RGB color watermark in color image using the visual characteristics of wavelet coefficient was proposed. As a result, the PSNR value of image was varied depending on perceptual parameter, but we can obtain 32dB as a whole.

• 한국정보통신설비학회:학술대회논문집
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• 한국정보통신설비학회 2008년도 정보통신설비 학술대회
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• pp.257-261
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• 2008

The Digital Radio Mondiale (DRM) system is a digital broadcasting standard designed for use in the LF, MF and HF bands of the broadcasting bands below 30 MHz. The system provides both superior audio quality and improved user services / operability compared with existing AM transmissions. In this paper, we propose a variable point Prime Factor FFT design method for Digital Radio Mondiale (DRM) system. Proposed method processes a various size IFFT/FFT of Robustness Mode on DRM standard efficiently by composing Radix-Prime Factor FFT Processing Unit of form similar to Radix-4 by insertion of a variable Prime Factor Twiddle Factor and Garbage data. So, we improved limitation that cannot process 112/176/256/288 FFT of each mode of DRM system with a existent Radix Processor and increase memory size and memory access time for IFFT/FFT processing by software processing in case of implementation with a existent high speed DSP.