• Proceedings of the IEEK Conference
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• summer
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• pp.199-203
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• 2004

In this paper, we only describe the digital block of two-channel 18-bit analog-to-digital (A/D) converter employing sigma-delta method and xl28 decimation. The device contains two fourth comb filters with 1-bit input from sigma­delta modulator. each followed by a digital half band FIR(Finite Impulse Response) filters. The external analog sigma-delta modulators are sampled at 6.144MHz and the digital words are output at 48kHz. The fourth-order comb filter has designed 3 types of ways for optimal power consumption and signal-to-noise ratio. The following 3 digital filters are designed with 12tap, 22tap and 116tap to meet the specification. These filters eliminate images of the base band audio signal that exist at multiples of the input sample rate. We also designed these filters with 8bit and 16bit filter coefficient to analysis signal-to-noise ratio and hardware complexity. It also included digital output interface block for I2S serial data protocol, test circuit and internal input vector generator. It is fabricated with 0.35um HYNIX standard CMOS cell library with 3.3V supply voltage and the chip size is 2000um by 2000um. The function and the performance have been verified using Verilog XL logic simulator and Matlab tool.

• Journal of Navigation and Port Research
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• v.28 no.3
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• pp.213-219
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• 2004

In this paper, a LNA(Low Noise Amplifier) has been developed, which is operating at L-band i.e., 1452∼1492 MHz for satellite DAB(Digital Audio Brcadcasting) receiver. The LNA is designed to improve input and output reflection coefficient and VSWR(Voltage Standing Wave Ratio) by balanced amplifier. The LNA consists of low noise amplification stage and gain amplification stage, which make a using of GaAs FET ATF-10136 and VNA-25 respectively, and is fabricated by hybrid method. To supply most suitable voltage and current, active bias circuit is designed Active biasing offers the advantage that variations in $V_P$ and $I_{DSS}$ will not necessitate a change in either the source or drain resistor value for a given bias condition. The active bias network automatically sets $V_{gs}$ for the desired drain voltage and drain current. The LNA is fabricated on FR-4 substrate with RF circuit and bias circuit, and integrated in aluminum housing. As a reults, the characteristics of the LNA implemented more than 32 dB in gain. 0.2 dB in gain flatness. lower than 0.95 dB in noise figure, 1.28 and 1.43 each input and output VSWR, and -13 dBm in $P_{1dB}$.

• Proceedings of the Korean Institute of IIIuminating and Electrical Installation Engineers Conference
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• 2007.05a
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• pp.400-403
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• 2007

This research describes how the class D amplifiers with power efficiency are designed and implemented for the PA audio systems. The configuration that makes use of the class D amplifier properties depends strongly on their applications. Thus in this paper the characteristics of the 2-level and 3-level PWM are analysed and the circuit implementation for them is presented. Using the proposed methods, they are designed and simulated for the further investigation. Test(Simulation) results present the improved performance that shows the satisfactory operations in controlling the PWM to the input signals.

• Electronics and Telecommunications Trends
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• v.27 no.6
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• pp.49-56
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• 2012

본고는 3DTV, UHDTV(Ultra High Definition Television)와 같은 실감방송 환경에서 실감 오디오 서비스를 제공하기 위한 초다채널 오디오 기술의 최근 연구 및 개발 동향을 소개한다. 스테레오와 5.1 채널로 대표되는 기존의 오디오 기술은 2차원 평면상에서만 음장을 형성할 수 있다는 표현의 한계를 가지고 있다. 3D 영화의 성공과 UHDTV로 대표되는 초고화질 비디오와 부합하기 위해서는 오디오도 3차원 공간상에서 표현되어야 하며 이를 위해서는 필연적으로 출력채널 수가 증가하여야 한다. 이러한 초다채널 오디오는 22.2 채널과 같은 대용량의 오디오 데이터를 압축하는 기술뿐만 아니라 다양한 오디오 출력 환경에 적응적으로 오디오 콘텐츠를 표현하는 기술에 대한 연구/개발이 필요하다.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2011.07a
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• pp.204-207
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• 2011

본 논문에서는 3DTV 의 입체감 향상을 위한 3D 음향의 원근감 재현 기술을 제안한다. 먼저, 3D 영상 객체의 깊이를 추출하고 영상 객체의 깊이에 따라 오디오 객체의 거리감을 조절한다. 오디오 거리감 재현을 위해 필요한 오디오 깊이 인자는 3D 영상의 좌/우 이미지의 차이 정보로부터 오디오에 맞도록 비선형 변환을 통해 구해진다. 3D 오디오 재현 알고리즘은 기존의 서라운드 입체음향 기술과 원근감 재현 기술로 구성된다. 원근감 재현 기술은 추정된 오디오 깊이 인자에 따라 신호크기, 초기 반사음, 근거리 머리전달함수, 위상 제어를 통해서 구현된다. 특히, 3D 영상 객체가 화면 앞으로 튀어 나올 때 소리도 튀어나오도록 함으로써 3D 영상 객체와 연동되는 입체 음향을 효과를 통해 3D 방송 시청 시 오디오/비디오 입체감을 향상시켜준다. 상용화된 3DTV 를 활용하여 음질 평가 전문가들의 주관 청취 평가를 통해 제안한 원근감 재현 기술이 3D 방송 시청에 적합함을 검증한다.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2003.11a
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• pp.69-72
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• 2003

This paper presents a signal processing IC for digital HDTV, which is designed to operate in bunt-in HDW or in HD-set-top Box. The chip supports de-multiplexing an ISO/IEC 13818-1 MPEG-2 TS stream. It decodes MPEG-2 MP@HL video bitstream, and provides high-quality scaled video for display on HDTV monitor. The chip consists of ARM7TDMI for TS-Demux, PCI interface, Audio interface, MPEG2 MP@HL video decoder Display processor, Graphic processor, Memory controller, Audio int3face, Smart Card interface and UART. It is fabricated using Sam sung's 0.18-um and the package of 492-pin BGA is used.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2011.05a
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• pp.611-614
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• 2011

최근 멀티미디어 기술의 발달, 특히 음향 기술의 급격한 발달과 더불어 고품질 오디오에 대한 요구와 함께보다 현실감 있는 오디오를 재생하기 위한 실감 오디오기술 개발이 요구되고 있다. 이러한 요구를 만족시키기 위해 사용자의 가상현실 및 증강현실에서 실감나는 오디오 효과를 제공해 줄 수 있는 3차원 오디오에 대한 연구가 활발히 진행되고 있다. 본 논문에서는 증강현실에서 좀 더 나은 오디오 기술을 적용하여 실감나는 오디오 효과를 제공해 줄 수 있는 방법을 연구하고자 하였다. 연구한 내용은 가상세계와 실제세계의 현실감을 제공하기 위하여 마커 위에 띄워진 3D 모델의 움직임에 따라서 움직임에 맞는 사운드. 즉, 거리, 각도 등의 변화에 따른 사운드의 크기 및 피치 변화를 줄 수 있도록 하였다.

• Electronics and Telecommunications Trends
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• v.33 no.5
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• pp.103-110
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• 2018

Studies on artificial intelligence have been developed for the past couple of decades. After a few periods of prosperity and recession, a new machine learning method, so-called Deep Learning, has been introduced. This is the result of high-quality big- data, an increase in computing power, and the development of new algorithms. The main targets for deep learning are 1D audio and 2D images. The application domain is being extended from a discriminative model, such as classification/segmentation, to a generative model. Currently, deep learning is used for processing 3D data. However, unlike 2D, it is not easy to acquire 3D learning data. Although low-cost 3D data acquisition sensors have become more popular owing to advances in 3D vision technology, the generation/acquisition of 3D data remains a very difficult problem. Moreover, it is not easy to directly apply an existing network model, such as a convolution network, owing to the variety of 3D data representations. In this paper, we summarize the 3D deep learning technology that have started to be developed within the last 2 years.

• The Journal of the Acoustical Society of Korea
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• v.28 no.7
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• pp.589-598
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• 2009

As mobile devices become multi-functional, and converge into a single platform, there is a strong need for a codec that is able to provide consistent quality for speech and music content MPEG-D USAC standardization activities started at the 82nd MPEG meeting with a CfP and approved WD3 at the 88th MPEG meeting. MPEG-D USAC is converged technology of AMR-WB+ and HE-AAC V2. Specifically, USAC utilizes three core codecs (AAC ACELP and TCX) for low frequency regions, SBR for high frequency regions and the MPEG Surround tool for stereo information. USAC can provide consistent sound quality for both speech and music content and can be applied to various applications such as multi-media download to mobile device Digital radio Mobile TV and audio books.

• The Journal of the Acoustical Society of Korea
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• v.28 no.7
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• pp.653-660
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• 2009

This paper proposes a robust audio watermarking algorithm for copyright protection and improvement of sound quality after embedding a watermark into an original sound. The proposed method computes the FFT (fast Fourier transform) of the original sound signal and divides the spectrum into a subbands. Then, it is necessary to calculate the energy of each subband and sort n subbands in descending order corresponding to its power. After calculating the energy we choose k subbands in sorted order and find p peaks in each selected subbands, and then embed a length m watermark around the p peaks. When the listeners hear the watermarked sound, they do not recognize any distortions. Furthermore, the proposed method is robust as much as Cox's method to MP3 compression, cropping, FFT echo attacks. In addition to this, the experimental results show that the proposed method is generally 10 dB higher than Cox's method in SNR (signal-to-noise ratio) aspect.