• The Journal of Korean Institute of Communications and Information Sciences
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• v.27 no.8B
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• pp.809-818
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• 2002

In this paper, we propose a fast cell search algorithm keeping compatible with the standard cell search algorithm of the WCDMA system. The proposed algorithm can acquire the synchronization of slot and frame times, and the code group identification using only one synchronization channel while the standard algorithm employs two synchronization channels called P-SCH and S-SCH. The proposed synchronization channel structure is the same as the P-SCH structure of the WCDMA system. However, the P-SCH is modulated with a specific polarization code, which is one element of new code group codes. The proposed algorithm can reduce both the BS' transmission power and the complexity of receiver as compared with the conventional one since only on synchronization channel is used. It is shown through the computer simulation that the proposed algorithm yields a significant improvement in terms of cell search time compared with the standard especially in low SNR environments.

• International Journal of Knowledge Content Development & Technology
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• v.7 no.1
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• pp.57-78
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• 2017

There are several statistical classification algorithms available for land use/land cover classification. However, each has a certain bias or compromise. Some methods like the parallel piped approach in supervised classification, cannot classify continuous regions within a feature. On the other hand, while unsupervised classification method takes maximum advantage of spectral variability in an image, the maximally separable clusters in spectral space may not do much for our perception of important classes in a given study area. In this research, the output of an ANN algorithm was compared with the Possibilistic c-Means an improvement of the fuzzy c-Means on both moderate resolutions Landsat8 and a high resolution Formosat 2 images. The Formosat 2 image comes with an 8m spectral resolution on the multispectral data. This multispectral image data was resampled to 10m in order to maintain a uniform ratio of 1:3 against Landsat 8 image. Six classes were chosen for analysis including: Dense forest, eucalyptus, water, grassland, wheat and riverine sand. Using a standard false color composite (FCC), the six features reflected differently in the infrared region with wheat producing the brightest pixel values. Signature collection per class was therefore easily obtained for all classifications. The output of both ANN and FCM, were analyzed separately for accuracy and an error matrix generated to assess the quality and accuracy of the classification algorithms. When you compare the results of the two methods on a per-class-basis, ANN had a crisper output compared to PCM which yielded clusters with pixels especially on the moderate resolution Landsat 8 imagery.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2010.05a
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• pp.798-801
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• 2010

This paper describes an efficient architecture for LDPC(Low-Density Parity Check) decoder, which supports three block lengths (648, 1,296, 1,944) of IEEE 802.11n standard. To minimize hardware complexity, the min-sum algorithm and block-serial layered structure are adopted in DFU(Decoding Function Unit) which is a main functional block in LDPC decoder. The optimized H-ROM structure for multi block lengths reduces the ROM size by 42% as compared to the conventional method. Also, pipelined memory read/write scheme for inter-layer DFU operations is proposed for an optimized operation of LDPC decoder.

• The Transactions of the Korean Institute of Electrical Engineers
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• v.35 no.2
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• pp.47-54
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• 1986

Hardware implementation of a 32-channel MFC digital receiver has not been easy and simple, because it requires real time processing of PCM data. In this paper, we introduce a method of designing an MFC digital receiver compactly by the channel distribution method. We have implemented the MFC digital receiver to process many cnannels by distributing channels of the TDM input data directly to the commercial digital signal processor chips(NEC uPD7720), and by carrying out the modified Goertzel Algorithm. The design of low cost, reliable, high speed, and compact MFC receiver will be shown.

• Proceedings of the Acoustical Society of Korea Conference
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• 1998.08a
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• pp.438-441
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• 1998

최근 미국 내 표준안으로서 많이 사용되고 있는 AC-3 오디오 알고리듬은 그 복잡성으로 인하여 실시간 구현을 위해선 프로세서로 구현하는 것이 적합하다. AC-3 복호화 알고리듬은 많은 부분이 실수연산으로 이루어져 있으므로 소수점을 고려한 연산이 필요한데, 프로세서로 구현할 때는 적은 비용과 빠른 속도로 실수연산을 수행하기 위해서 부동소수점보다는 고정소수점 연산이 유리하다. 그러나 고정소수점 연산시 발생하는 유한 단어길이 효과로 인하여 양자화 오차가 발생하므로 복호화된 오디오 신호의 음질저하를 최소화하기 위해서는 최적화가 필요하다. 본 논문에서는 AC-3 복호화 알고리듬의 부분별 양자화 오차를 분석하고 그 결과 가장 많은 오차를 발생시키는 역 TDAC 변환의 오차를 최적화하였다. Fast TDAC 변환이 FFT로 이루어져 있으므로 고정 소수점 연산시 오차가 적은 FFT 구조를 제안하였다. 제안된 구조를 사용하여 AC-3 고정소수점 복호화기를 C 언어를 사용하여 구현하였으며, AC-3 부동소수점 복호화기와 최종 PCM을 비교하여 그 성능을 평가하였다.

• Proceedings of the KIEE Conference
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• 2002.11c
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• pp.577-580
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• 2002

본 논문에서는 $\Sigma\Delta$ 변조된 입력신호를 갖는 적응필터의 수렴특성 및 연산량을 향상시키는 방안을 제시하였다. 하드웨어측면에서 효율적인 해상도를 내는 $\Sigma\Delta$ 변조기는 중저파 대역의 신호를 처리하는데 널리 사용되고 있다. $\Sigma\Delta$ 변조신호는 항상 $\pm$1의 값만을 갖기 때문에, 사인알고리즘을 사용하는 적용필터와 효율적으로 결합될 수 있다. 하지만 PCM 신호에 비하여 $\Sigma\Delta$ 변조신호의 상대적인 길이가 길어 이를 처리하는 적응필터의 길이가 증가하고 이에 따른 연산량도 증가하고, 아울러 사인 알고리즘 자체가 갖는 수렴속도의 문제점 때문에 이러한 결합은 불안정한 수렴 특성을 보이게 된다. 본 연구에서는 $\Sigma\Delta$ 변조된 입력신호에 대하여 웨이블릿 변환을 적용한 변환영역 적응필터를 설계하였으며, 필터계수의 일부분만을 주기적으로 갱신함으로써 연산량을 줄이는 방안과 수렴속도의 향상됨을 시스템 식별의 응용 예를 통하여 검증하였다.

• Proceedings of the KIEE Conference
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• 2002.07d
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• pp.2586-2588
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• 2002

본 논문에서는 $\Sigma\Delta$ 변조된 입력신호를 갖는 적응필터의 수렴특성을 연구하여 향상 방안을 제시하였다. 하드웨어적인 측면에서 효율적인 해상도를 내는 $\Sigma\Delta$ 변조기는 중저주파 대역의 신호를 처리하는데 널리 사용되고 있다. $\Sigma\Delta$ 변조신호는 항상 $\pm1$의 값만을 갖기 때문에, 사인 알고리즘을 사용하는 적응필터와 효율적으로 결합될 수 있다. 하지만, PCM 신호에 대비하여 $\Sigma\Delta$ 변조 신호의 상대적인 길이가 길어 이를 처리하는 적응필터의 길이가 증가하고, 아울러 사인 알고리즘 자체가 갖는 수렴속도의 문제점 때문에 이러한 결합은 불안정한 수렴 특성을 보이게 된다. 본 연구에서는 $\Sigma\Delta$ 변조된 입력신호에 대하여 웨이블릿 변환을 적용한 변환영역 적응필터를 설계하였으며, 수렴속도가 향상됨을 시스템 식별의 응용예를 통하여 검증하였다.

• Proceedings of the KIEE Conference
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• 1998.07g
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• pp.2414-2416
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• 1998

This paper presents real-time implementation of speech vocoder for PSTN video telephony using ITU G.723 16Kbps ADPCM algorithm. The ADPCM encoder accepts 8-bit PCM compressed signals and expends it to a 14-bit-per-sample. The predicted values are subtracted from encoded signals to produce difference signals. Adaptive quantization is performed on the difference signal to produce a 2-bit, output for transmission over the channel. Computer simulations and experiments were performed to evaluate the performance of the speech vocoder.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2010.05a
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• pp.771-774
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• 2010

In this paper, BER performance of LDPC(Low-Density Parity-Check) decoder for WiMAX is analyzed, and optimal design conditions of LDPC decoder are derived. The min-sum LDPC decoding algorithm which is based on an approximation of LLR sum-product algorithm is modeled and simulated by Matlab, and it is analyzed that the effects of LLR approximation bit-width and maximum iteration cycles on the bit error rate(BER) performance of LDCP decoder. The parity check matrix for IEEE 802.16e standard which has block length of 2304 and code rate of 1/2 is used, and AWGN channel with QPSK modulation is assumed. The simulation results show that optimal BER performance is achieved for 7 iteration cycles and LLR bit-width of (8,6).

• The Journal of Korean Institute of Communications and Information Sciences
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• v.26 no.11A
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• pp.1908-1924
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• 2001

Interoperability and inter-working in the various network and media environment with different technology background is very important to enlarge the opportunity of service access and to increase the competitive power of service. The ITU-T and advanced counties are planning ahead for provision of GII enabling user to access advanced global communication services supporting multimedia communication applications, embracing all modes of information. In this paper, we especially forced the heterogeneity of end user applications for multimedia networking. The heterogeneity has several technical aspects, like different medium access methods, heterogeneous coding algorithms for audio-visual data and so on. Among these elements, we have been itemized bit rate shaping algorithm on the compressed moving video. Previous manipulations of video has been done on the uncompressed signal domain. That is, compressed video should be converted to linear PCM signal. To do such a procedures, we should decode, manipulate and then encode the video to compressed signal once again. The traditional approach for processing the video signa1 has several critical weak points, requiring complexity to implement, degradation of image quality and large processing delay. The bit rate shaping algorithm proposed in this paper process the manipulation of moving video on the completely compressed domain to cope with above deficit. With this algorithms. we could realized efficient video bit rate shaping and the result of software simulation shows that this method has significant advantage than that of pixel oriented algorithms.