• Journal of the Institute of Electronics Engineers of Korea SD
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• v.44 no.11
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• pp.1-8
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• 2007

A wideband continuous-time sigma delta modulator for wireless application is implemented in 130nm CMOS. The SNR for small input signal is improved using a proposed adaptive quantizer which can effectively scale the quantization level. The modulator comprises a second-order loop filter for low power consumption, 4-bit quantizer and DAC for low jitter sensitivity and high linearity. Designed circuit achieves peak SNR of 51.36B with 10MHz signal Bandwidth and 320MHz sampling frequency dissipating 30mW.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 1999.11a
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• pp.199-203
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• 1999

This paper is on the DSP system design and implementation for real time MPEG-2 AAC multichannel audio, and MPEG-4 object oriented audio coding. This DSP system employs two DSPs of the state of the art TMS320C6701, developed by TI semiconductor. DSP board has PCI interface for downloading application program and control the system. DSP board was designed to use for both encoder and decoder, by setting several switches. The system contains external input and output box also, for A/D and D/A conversion for eight channel audio. The input box converts multi channel digital audio to ADI format, that provides serial interface for eight channel digital audio. And the output box converts ADI format signal to multi channel audio. Through this ADI interface, DSP boards can be connected to input, output box. Implemented DSP system was tested for integration with MPEG-2 AAC encoder and decoder S/W. Currently the DSP system performs realtime AAC 4-channel audio encoding with two DSPs, and 8-channel decoding with one DSP.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.5C
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• pp.654-654
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• 2004

본 논문에서는 이산웨이블렛 변환을 이용한 영상 압축 프로세서를 하드웨어로 구현하였다. 웨이블렛 변환을 위하여 필터뱅크 및 피라미드 알고리즘을 이용하였고 각 필터들은 FIR 필터로 구현하였다. 병렬구조로 이루어져 동일 클럭 싸이클에서 하이패스와 로패스를 동시에 수행함으로써 속도를 향상시킬 뿐 아니라 QMF 특성을 이용하여 DWT 연산에 필요한 승산기의 수를 절반으로 줄임으로써 하드웨어 크기를 줄이고 이용효율 또한 높일 수 있다. 다중 해상도 분해 시 필요한 메모리 컨트롤러를 하드웨어로 구현하여 DWT 계산이 수행되므로 이 융자는 단순한 파라메터 입력만으로 효과적인 압축율을 얻을 수 있도록 구조적으로 설계하였다. 실시간 영상압축 프로세서의 성능 예측을 위하여 MATLAB을 통하여 시뮬레이션 하였고, VHDL을 이용하여 각 모듈들을 설계하였다. 설계한 영상압축기는 Leonaro-Spectrum에서 합성하였고, ALTERA FLEX10KE(EPF10K100 EFC256) FPGA에 이식하여 하드웨어적으로 동작을 검증하였다. 설계된 부호화기는 512×512 Woman 영상에 대하여 33㏈의 PSNR값을 갖는다. 그리고 설계된 프로세서를 FPGA 구현 시 35㎒에서 정상적으로 동작한다.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.7 no.8
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• pp.1674-1683
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• 2003

The performance of a packet data multiplexer with stop­and­wait ARQ protocols under Markovian interruption is considered in this work It is assumed that the input process, into the system is Poisson process, and that the output channel is divided into a series of time slots and a data packet can be transmitted in a slot time. In this system the round­trip propagation delay is defined to be the frame time. It is modeled that the output channel can be blocked by some Markevian interruption, whose state change between the blocking and non­-blocking states is given by Markov process. The overall system has been analyzed by constructing a relationship, taking the Markovian interruption into account, about the buffer behavior between the successive frames of slots. The validity of this analytical results has been verified by computer simulation.

• Journal of Korean Institute of Industrial Engineers
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• v.25 no.2
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• pp.162-172
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• 1999

For VBR video traffic, Motion-Picture Experts Group(MPEG) coding algorithm was adopted as the standard coding algorithm by International Telecommunication Union(ITU). In this paper, we propose a traffic model of an MPEG coded video traffic in frame level and cell level, and develop an approximate model for evaluating performance of a ATM multiplexer with homogeneous MPEG video sources by considering burst-level variation of aggregated traffics. For homogeneous MPEG video traffics which are frame-synchronized, the performance of the ATM multiplexer is influenced by source correlation at the multiplexing time. When sources are highly correlated, we decompose the aggregated cell streams by the frame-type and model multiplexing process during a frame time as n*D/D/1/K queueing model and suggest an approximate method for obtaining CLP of the ATM multiplexer. In the case that sources are highly correlated, the solution has the meaning of the upper bounds of performance of the ATM multiplexer. For the verification of our model, we compare the solution of our model with simulation resets. As the number of sources increases. The CLP obtained from our model approaches to simulation results, and gives upper bounds of simulation results.

• Journal of Broadcast Engineering
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• v.25 no.2
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• pp.252-264
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• 2020

Korea has launched the world-wide first terrestrial UHD broadcast services on May 31, 2017. While the existing HDTV broadcast services use MPEG-2 TS (Tranport Stream) standard for multiplexing and delivering compressed media with additional data, the terrestrial UHD broadcast services use MMT (MPEG Media Transport) standard, which is the next-generation standard beyond MPEG-2 TS. However, the production cost of UHD contents is so high that only a part of the total broadcast time is filled with UHD contents and the UHD time portion is planned to be gradually increased. On the other hand, the ATSC 3.0 standard that uses MMT is not yet used in full-fledged broadcast services in North America. Hence MMT broadcast equipment is still at an early stage with high prices. In this paper we implemented a multi-thread software running on an ordinary PC that can be utilized to realize a low-cost converter that converts the output of an existing MPEG-2 TS multiplexer to an MMTP (MMT Protocol) packet stream. We also verified the functionality of the software through experiments.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.48 no.5
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• pp.129-136
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• 2011

This paper presents the coding method that have multi-mode and efficiency of audio codecs using the feature of audio signal. Recently, the developed extension super-wideband codec based on G.718 wideband divides two mode between Generic and Sinusiodal. So codec efficently encode audio signal exist in super-wideband. But the codec is not as efficent coding for harmonic component of wind instrument and string instrument and individual-Line component of percussion instrument. The proposed method are modeling and encoding multiple pitch and individual-line feature using multi mode coding. For the performance evaluation, we used SNR in MDCT domain for objective test and MUSHRA test for subjective test. As a result, the performance of SNR and MUSHRA test of the proposed method have better performance than the G.718 super-wideband codec.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.22 no.9
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• pp.2035-2042
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• 1997

We consider the design of soft decision functions for each stage of multistage detection for coherent demodulation in an asynchronous code-division multiple-access(CDMA) system. In particular, the sigmoid function, which is shown to be optimal under the mean square error(MSE) criterion, andmultilevel quantizers that best approximate the sigmoid function are derived. At each stage of multistage detection, the parameters of these decision functions are adjusted depending on estimated input statistics. Computer simulation results demonstrate that multistage detectors employing these soft decision functions perform considerably better than those with hard decision.

• Journal of The Institute of Information and Telecommunication Facilities Engineering
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• v.4 no.1
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• pp.37-44
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• 2005

본 논문에서는 멀티 코드 CDMA(Code Division Multiple Access)시스템에서 고속의 멀티미디어 서비스를 지원하기 위해 정보 데이터열을 QAM(Quadrature Amplitude Modulation) 부호화하는 방법을 제안한다. 송신 구조는 입력 비트를 정진폭 다중부호 이진직교 변조(Constant amplitude code biorthogonal modulation: 이하 CACB라 칭함)로 부호화하여 전송 심볼의 크기를 일정하게 하고 이렇게 이진화된 신호를 QAM 부호화하여 전송속도를 높인다. 복조기의 구조는 수신된 신호로부터 QAM 연판정기 블록을 거쳐 생성된 신호를 CACB 복호화기를 거쳐 데이터를 복조해 낸다. 제안된 시스템은 기존의 멀티 코드 방식에 비해 대역폭 효율을 크게 개선시킬 수 있어 전송 속도를 많이 향상시킬 수 있고, 멀티 코드 방식을 기반으로 하기 때문에 디지털 가전기기나 3세대 이동통신과, WPAN과 관련된 무선 네트워크 응용과 같은 미래의 고속의 무선 멀티미디어 서비스를 지원하기에 적합하다.

• Journal of the Korean Institute of Telematics and Electronics C
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• v.36C no.7
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• pp.36-45
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• 1999

In order to control the flow of traffics in ATM networks and optimize the usage of network resources, an efficient control mechanism is necessary to cope with congestion and prevent the degradation of network performance caused by congestion. In this paper, Buffered Leaky Bucket which applies the same control scheme to a variety of traffics requiring the different QoS(Quality of Service) and Neural Networks lead to the effective buffer utilization and QoS enhancement in aspects of cell loss rate and mean transfer delay. And the cell scheduling algorithms such as DWRR and DWEDF for multiplexing the incoming traffics are enhanced to get the better fair delay. The network congestion information from cell scheduler is used to control the predicted traffic loss rate of Neural Leaky Bucket, and token generation rate and buffer threshold are changed by the predicted values. The prediction of traffic loss rate by neural networks can enhance efficiency in controlling the cell loss rate and cell transfer delay of next incoming cells and also be applied for other traffic controlling schemes. Computer simulation results performed for random cell generation and traffic prediction show that QoSs of the various kinds of traffcis are increased.