• Proceedings of the IEEK Conference
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• 2006.06a
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• pp.511-512
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• 2006

Three stage(CIC-FIR-FIR) decimation filter in delta-sigma A/D converter for audio is designed. A canonical signed digit(CSD) code method is used to minimize area of multipliers. This filter is designed in 0.25um CMOS process and incorporates $1.36\;mm^2$ of active area. Measured results show that this decimation filter is suitable for digital audio A/D converters.

• Journal of IKEEE
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• v.14 no.4
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• pp.297-304
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• 2010

This paper presents a link layer design of DisplayPort interface with a state machine based on packet processing. The DisplayPort link layer provides isochronous video/audio transport service, link service, and device service. The merged video, audio main link, and AUX channel controller are implemented with 7,648 LUTs(Loop Up Tables), 6020 register, and 821,760 of block memory bits synthesized using a FPGA board and it operates at 203.32MHz.

• Journal of the Korea Society of Computer and Information
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• v.3 no.4
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• pp.10-18
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• 1998

Video information is an important component of multimedia system such as Digital Library. World-Wide Web(WWW) and Video-On-Demand(VOD) service system. It can support various types of information because of including audio-visual, spatial-temporal and semantics information. In addition, it requires the ability of retrieving the specific scene of video instead of entire retrieval of video document. Therefore, so as to support a variety of retrieval, this paper models metadata using video document structure information that consists of hierarchical structure, and designs database schema that can manipulate video document.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.9C
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• pp.851-859
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• 2003

최근 사용자들의 멀티채널 선호도는 급속도로 전파되고 있다. MPEG은 동영상 및 음향시스템의 데이터 압축기술을 제공하는데, 현재 각광을 받고있는 것이 디지털 오디오이다. MPEG 표준안은 MPEG-1오디오 알고리즘을 MPEG-2 알고리즘에 동일하게 사용해도 멀티채널 및 5.1채널 사운드륵 제공한다. MDCT(Modified Discrete Cosine Transform)는 TDAC(Time Domain Aliasing Cancellation)에 기반을 두고있는 변형이산 여현 변환을 나타낸 것이다. 본 논문에서는 오디오 부분의 핵심이라 할 수 있는 MDCT/IMDCT(Inverse MDCT) 알고리즘을 최적화하여 효율적인 알고리즘을 제안하였다. 그리고 연산과정에서 중복되는 영역을 묶음으로써 연산에 필요한 계수를 줄였다. 최적화 전에 비해 코사인 계수를 0.5%이하로 최적화하였고, 승산에서 0.098%, 가산에서 0.58% 효율을 보였다. 알고리즘 검증은 C언어를 사용하여 검증하였고, 최적화된 알고리즘을 적용하여 마이크로 프로그램 방식의 하드웨어 구조론 설계하였다.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.18 no.3
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• pp.177-183
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• 2018

Currently, digital signal processing systems are used extensively in image processing, audio processing, filtering, and equalizations, etc. In addition, the importance of DRAM, which has a great influence on the performance of an digital signal processor has been increased, making research on DRAM actively conducted in industry and academia. Therefore, it is important to have a more accurate DRAM model in order to obtain reliable results when evaluating the performance of a digital signal processor through simulation. In this paper, we developed a digital signal processor simulator capable of inter-working with a DRAM simulator. With the simulator, we analyzed the influence of the DRAM model which operates correctly on a cycle-by-cycle basis, on the performance of the digital signal processor by using the UTDSP digital signal benchmark.

• Journal of the Institute of Electronics Engineers of Korea SD
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• v.42 no.4 s.334
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• pp.61-68
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• 2005

A chip, which can fast encoder the audio data to AAC (Advanced Audio Coding) LC(Low Complexity) that is MPEG-2 audio standard, has been designed on the basis of a 32 bits DSP core and fabricated with 0.25um CMOS technology. At first, the various optimization methods for implementing the algerian are devised to reduce the memory size and calculation cycles. FFT(Fast Fourier Transform) hardware block is added to the DSP core to get the more reduction of the calculation cycles. The chips has the size of $7.20\times7.20 mm^2$ and about 830,000 equivalent gates, can carry out AAC encoding under 70MIPS(Million Instructions per Second).

• Journal of Broadcast Engineering
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• v.21 no.3
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• pp.357-368
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• 2016

In DAB+, HE-AAC v2 codec is applied for the fixed rate transmission of audio stream. In case that xHE-AAC codec including USAC, a more efficiency is expected when the variable frame is used in a given same bandwidth compared to the fixed frame transmission. For this to be realized, audio streams need to be multiplexed in a sub-channel before transmission, then a method is required to identify the border of each audio frames. In this paper, the toggled sync byte and additional identification field being sequentially placed between AU borders are proposed in order to deal with the AU border identification. In addition, the Reed-Solomon based error correction code which is compliant to DAB+ is proposed.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2014.06a
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• pp.157-159
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• 2014

In this paper, we introduce SlimSRP, an ultra-low-power digital signal processor (DSP) solution for mobile audio and voice applications. So far, application processors (APs) have taken charge of all the tasks in mobile devices. However, they have suffered from short battery life problems to deal with complex usage scenarios, such as always-on voice trigger with continuous audio playback. From extensive analysis of audio and voice application characteristics, SlimSRP is designed to relive the performance and power burden of APs. It employs three-issue VLIW architecture, and the major low-power and high-performance techniques include: (1) an optimized register-file architecture friendly for constants generation, (2) a powerful instruction set to reduce the number of register file accesses and (3) a unique instruction compression scheme that contributes to saved memory size and reduced cache miss. An implementation of SlimSRP runs at up to 200MHz and the logic occupies 95K NAND2 gates in Samsung 28LPP process. The experimental results demonstrate that a MP3 decoder application with a 128kbps 44.1kHz input can run at 5.1MHz and the logic consumes only 22uW/MHz.

• Journal of Broadcast Engineering
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• v.11 no.2 s.31
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• pp.200-206
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• 2006

This paper presents a preprocessing method to modify the input audio signals of a speech coder to obtain the finally enhanced signals at the decoder. For the purpose, we introduce the noise suppression (NS) scheme and the adaptive gain control (AGC) where an audio input and its coding error are considered as a noisy signal and a noise, respectively. The coding error is suppressed from the input and then the suppressed input is level aligned to the original input by the following AGC operation. Consequently, this preprocessing method makes the spectral energy of the music input redistributed all over the spectral domain so that the preprocessed music can be coded more effectively by the following coder. As an artifact, this procedure needs an additional encoding pass to calculate the coding error. However, it provides a generalized formulation applicable to a lot of existing speech coders. By preference listening tests, it was indicated that the proposed approach produces significant enhancements in the perceived music qualities.

• The Journal of The Korea Institute of Intelligent Transport Systems
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• v.2 no.2 s.3
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• pp.1-12
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• 2003

DAB(Digital audio broadcasting) is a next generation radio broadcasting system which provides CD quality audio, various data services and superior reception ability when moving. Also, it can show traffic informations and news literally or graphically. In this paper, we design and fabricate the DAB tuner for ITS service that follows Eureka-147 and ETSI 300 401 specifications. This small-sized tuner can be adopted to mny electronic equipments such as a Hi-Fi audio, DVD player, car audio system etc.. The overall performance of the tuner depends on a phase noise of VCO and the sensitivity of the receiving system is influenced by LNA, image rejection filter and channel selection filter. All our measurement results satisfy the specification for a DAB system with the return loss of 9dB, the noise figure of 6dB for both Band 111 and L-band and the sensitivity of -97dBm.