• 대한전기학회:학술대회논문집
• /
• 대한전기학회 2005년도 심포지엄 논문집 정보 및 제어부문
• /
• pp.170-172
• /
• 2005

The DSD(Direct Stream Digital) encoder, which is a standard for SACD(Super Audio Compact Disc) proposed by Sony and philips, use 1 bit representation with a sampling frequency of 2.8224 MHz (64 $\times$ 44.1 kHz). For multi-rate PCM (Pulse Code Modulation) input like as 48/96/192 kHz, a external sample-rate converter is necessary to the DSD encoder. This paper has been proposed a digital filter structure composed of sample-rate converter and interpolation filter for the DSD encoder with multi-rate (48/96/192 kHz) PCM input. without a external sample-rate converter.

• 한국고분자학회:학술대회논문집
• /
• 한국고분자학회 2006년도 IUPAC International Symposium on Advanced Polymers for Emerging Technologies
• /
• pp.214-214
• /
• 2006

The huge amounts of wastes were produced nowadays. Among the solid waste stream, the waste plastic portion is about 20 % in weight, over 50 % in volume. But the most of waste plastics were incinerated or land filled. Only a bit of waste plastics were recycled and reused. On the view point of current energy crisis, this will be an extravagance of beneficial resource. So we should consider the waste plastics as a beneficial raw material. Also need to develop the field of reusing and recycling of post-consumed plastic. In this context, present paper describes the methodology for enhancing or improving the specific properties by way of using inorganic filler waste. Also develop the product from these post-consumed plastic.

• ETRI Journal
• /
• 제34권1호
• /
• pp.106-109
• /
• 2012

In this letter, we propose a new prediction block filter that can reduce errors between the original and prediction blocks. The proposed filter adaptively adjusts filter coefficients by using the previously reconstructed adjacent blocks and their prediction blocks. Then, the filter is selectively applied to the current prediction block according to the rate-distortion optimization. Moreover, since the same filter coefficients can be derived in the decoder, they are not encoded into the bit-stream. The proposed method achieves a 4.65% bitrate saving on average compared with H.264/AVC.

• 한국정보처리학회논문지
• /
• 제7권8호
• /
• pp.2341-2348
• /
• 2000

현재 동영상 데이터를 이용한 압축영역에서의 특수효과 구현에 많은 관심이 모여지고 있다. 따라서 본 논문에서는 DCT (Discrete Cosine Transform)압축영역에서 다양한 효과를 위해 최소한의 디코더를 개발하였으며, 이를 이용하여 Wipe, Dissolve, Zoom등의 비디오 효과등을 DCT영역에서 제안하였다. 또한 구현한 디코더와 효과들의 확장성과 이식성이 용이하도록 COM기반의 DirectShow 형식으로 각 모듈을 필터화 하였다.

• 대한전기학회:학술대회논문집
• /
• 대한전기학회 2005년도 학술대회 논문집 정보 및 제어부문
• /
• pp.358-360
• /
• 2005

The DSD(Direct Stream Digital) encoder, which is a standard for SACD(Super Audio Compact Disc) proposed by Sony and philips, uses 1 bit representation with a sampling frequency of 2.8224MHz (64${\times}$44.1kHz). For multi-rate PCM (Pulse Code Modulation) input such as 8${\sim}$192kHz, a external sample-rate converter is necessary to the DSD encoder. This paper has been proposed a digital mter structure composed of sample-rate converter and interpolaton filter for the DSD encoder with multi-rate (8${\sim}$192kHz) PCM input, without a external sample-rate converter.

• 대한전자공학회:학술대회논문집
• /
• 대한전자공학회 2002년도 ITC-CSCC -2
• /
• pp.872-875
• /
• 2002

A digital watermarking technique applying psychoacoustic model for audio signal is proposed in this paper. In the watermarking scheme, the pseudo-random bit stream used as a watermark signal is embedded into the audio signal in both speech and music. The strength of the embedded signal is subject to the human auditory system in such a way that the disturbances on host audio signal are beyond the sensing of human ears. The experimental results show that the quality of the watermarked audio signal, in term of signal to noise ratio, can be improved up to 3.2 dB.

• 전자공학회논문지A
• /
• 제33A권5호
• /
• pp.180-187
• /
• 1996

This paper describes a post-layout simulation method using VHDL and C for verifying the architecture of antifuse-based FPGAs and the dedicated CAD system. An antifuse-based FPGA consists of programming circuitry including decoding logic, logic modules, segmented tracks, antifuses and I/O pads. The VHDL model which includes all these elements is used for logic verification and programming verification of the implemented circuit by reconstructing the logic circuit from the bit-stream generated from layout tool. The implemented circuit comprises of logic modules and routing networks. Since the routing delay of the complex networks is comparable to the delay of the logic module in the FPGA, the accurate post-layout simulation is essential to the FPGA system. In this paper, the C program calculates the delay of the routing netowrks using SPICE, elmore or horowitz delay models and the results feedback to the VHDL simulation. Critical path anc be found from this post-layout simulation results.

• ETRI Journal
• /
• 제18권4호
• /
• pp.229-244
• /
• 1997

Realization of the economical, reliable, and efficient ATM interface block becomes an important key to development of the ATM switching system when we consider new issues raised recently. In this paper, we summarize requirements for the ATM interface block and present the UNI (User Network Interface)/NNI (Network Node Interface) architecture to meet these requirements. We also evaluate the performance of the multiplexer adopting the various multiplexing schemes and service disciplines. For ATM UNI/NNI interface technologies, we have developed a new policing device using the priority encoding scheme. It can reduce the decision time for policing significantly. We have also designed a new spacer that can space out the clumped cell stream almost perfectly. This algorithm guarantees more than 99 % conformance to the negotiated peak cell rate. Finally, we propose the interface architecture for accommodation of the ABR (Available Bit Rate) transfer capability. The proposed structure that performs virtual source and virtual destination functions as well as a switch algorithm can efficiently accommodate the ABR service.

• ETRI Journal
• /
• 제42권3호
• /
• pp.446-458
• /
• 2020

In this study, an efficient scheme for hiding data directly in partially encrypted versions of high efficiency video coding (HEVC) videos is proposed. The content owner uses stream cipher to selectively encrypt some HEVC-CABAC bin strings in a format-compliant manner. Then, the data hider embeds the secret message into the encrypted HEVC videos using the specific coefficient modification technique. Consequently, it can be used in third-party computing environments (more generally, cloud computing). For security and privacy purposes, service providers cannot access the visual content of the host video. As the coefficient is only slightly modified, the quality of the decrypted video is satisfactory. The encrypted and marked bitstreams meet the requirements of format compatibility, and have the same bit rate. At the receiving end, data extraction can be performed in the encrypted domain or decrypted domain that can be adapted to different application scenarios. Several standard video sequences with different resolutions and contents have been used for experimental evaluation.

• 한국소음진동공학회논문집
• /
• 제18권6호
• /
• pp.612-618
• /
• 2008

Microphone array beamforming method has been recognized as an important aeroacoustic research field and become a standard technique in localizing sound sources. This method also used in flight acoustic measurement, and especially, it is very useful when measure sounds inside the wind tunnel. In measuring sound which is inside the wind tunnel by traditional beamforming method, there are some errors caused by airstream. The speed and the propagation path of the sound changes as it travel through the airstream. This makes the error which the position of sound is changed a little bit to the down stream direction. In this paper, validation test has made about the correction equation for this wind effects of previous researches. And beamforming including shear layer correction was performed about a sound source in the anechoic open-jet wind tunnel.