• Journal of the Institute of Electronics Engineers of Korea SP
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• v.46 no.2
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• pp.48-56
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• 2009

We propose a fast intra mode skip decision algorithm for H.264/AVC video encoding. Although newly added MB encoding algorithms based on various prediction methods increase compression ratio, they require a significant increase in the computational complexity because we calculate rate-distortion(RD) cost for all possible MB coding modes and then choose the best one. In this paper, we propose a fast mode decision algorithm based on an adaptive motion vector map(AMVM) method for H.264/AVC video encoding to reduce the processing time for the inter frame. We verify that the proposed algorithm generates generally good performances in PSNR, bit rates, and processing time.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2016.11a
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• pp.60-62
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• 2016

이동통신 시스템의 OFDM(Othogonal frequency division multiplexing) 신호는 큰 PAPR(Peak to Average Power Ratio)을 가지기 때문에 비선형 특성을 가지는 전력 증폭기의 효율 감소를 가져온다. 이러한 전력 증폭기의 비선형 특성을 개선하여 효율을 증가시키기 위해서 전력 증폭기의 역 특성을 가지는 디지털 전치 왜곡기가 이용된다. 본 논문에서는 제곱근 근사를 이용한 Look-up Table(LUT) 기반의 디지털 전치왜곡(Digital Pre-Distortion :DPD) 기법을 제안한다. 제안하는 방식은 복소 이득(Complex Gain) LUT 구조에서 입력신호의 크기를 구할 때, 기존의 테이블을 이용하여 제곱근 연산을 하는 방식보다 좋은 성능을 내면서 근사를 위한 테이블의 메모리를 필요로 하지 않는다. 또한 간단한 쉬프트 연산 등을 이용하므로 DSP 또는 MCU 기반의 DPD를 구현할 때 간단하게 구현 될 수 있다는 장점을 갖는다. 컴퓨터 모의실험을 통해 제안하는 제곱근 근사방식을 이용한 DPD와 기존의 방식을 사용한 DPD를 비교함으로써 제안하는 방식이 기존 방식보다 좋은 성능을 내면서도 보다 효율적으로 구현될 수 있음을 검증하였다.

• The Transactions of The Korean Institute of Electrical Engineers
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• v.64 no.12
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• pp.1756-1760
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• 2015

This paper focuses on a priori signal to noise ratio (SNR) estimation method for the speech enhancement. There are many researches for speech enhancement with several ambient noise cancellation methods. The method based on spectral subtraction (SS) which is widely used in noise reduction has a trade-off between the performance and the distortion of the signals. So the need of adaptive method like an estimated a priori SNR being able to making a high performance and low distortion is increasing. The decision directed (DD) approach is used to determine a priori SNR in noisy speech signals. A priori SNR is estimated by using only the magnitude components and consequently follows a posteriori SNR with one frame delay. We propose a modified a priori SNR estimator and the weighted rational transfer function for speech enhancement with wind noises. The experimental result shows the performance of our proposed estimator is better Perceptual Evaluation of Speech Quality scores (PESQ, ITU-T P.862) compare to the conventional DD approach-based systems and different noise reduction methods.

• Journal of Korea Multimedia Society
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• v.18 no.4
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• pp.437-442
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• 2015

Voice activity detection is very important process that voice activity separated form noisy speech signal for speech enhance. Over the past few years, many studies have been made on voice activity detection, but it has poor performance in low signal to noise ratio environment or fickle noise such as car noise. In this paper, it proposed new voice activity detection algorithm using ensemble variance based on wavelet band entropy and soft thresholding method. We conduct a survey in a lot of signal to noise ratio environment of car noise to evaluate performance of the proposed algorithm and confirmed performance of the proposed algorithm.

• Proceedings of the KIEE Conference
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• 2006.04b
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• pp.121-123
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• 2006

In this paper, the notch of teeth and skew of permanent magnet are used to reduce the detent force caused by slot-teeth structure. Also, the shape of permanent magnet is optimized to reduce the detent forceowing to flux hannonics components of permanent magnet. As a result, thrust is decrease about 2[%]. But, the distortion ratio of thrust is decreased from 1.04[%] to 0.75[%]. And, the ripple ratio of thrust is decreased from 2.6[%] to 1.65[%].

• Journal of electromagnetic engineering and science
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• v.3 no.1
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• pp.7-11
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• 2003

CDMA(code division multiple access) has very large peak to average power ratio(PAR) and behave as noise-like wide band digital signals with 1.2288 Mbps transmission rate. For signals with high PAR like CDMA, it is reasonable to prescribe occupied bandwidth(OBW) as average occupied bandwidth. Bandwidth measurements of CDMA signals at remote site are affected by co-channel and adjacent channel interference from adjacent CDMA base station, distortion of signal by fading effect, spurious emission and environment noises. In this study, we have compared OBW measurements in an on-air environment with those measured in a base station using adjacent channel leakage ratio(ACLR) as a reference measurement factor. As results of analysis, the OBW at ACLR$\geq$35 ㏈ shows nearly same statistical characteristics regardless of the measurement locations and environments.

• Proceedings of the KIEE Conference
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• 2011.07a
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• pp.1876-1877
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• 2011

In this paper, a winding fault diagnosis method base on MCSA(Motor Current Signature Analysis) for BLDC motor is proposed. This method is programmed by LabVIEW for winding fault diagnosis. For winding fault diagnosis, two types of winding fault(shorted turn at one pole, shorted turn at two pole in same phase) are put intentionally in on phase. The motor current is collected by hole sensor, and transformed by the Park's transform, and then the Park's Vector Pattern are obtained, Usually this pattern is formed an ellipse, so a proper threshold value of distortion ratio(the ratio of the shortest axis and longest axis of ellipse) is suggested for winding faults diagnosis.

• Proceedings of the Acoustical Society of Korea Conference
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• 1998.06c
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• pp.147-150
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• 1998

In the case of speech synthesis, the waveform coding method with high quality is mainly used to the synthesis by analysis. Because the parameters of this coding method are not classified as both excitation and vocal tract parameters, it is difficult to apply the waveform coding method to the synthesis by rule. Thus, in order to apply the waveform coding method to the synthesis by rule, a pitch alteration is required for the prosody control. In the speech synthesis method by the conventional PSOLA technique, applying symmetric window function to asymmetric speech waveform, it occurs the unbalance phenomenon of energy according to the overlapped degree of pitch interval adjustment. In this paper to overcome the unbalance phenomenon of energy, we proposed a new method that can convert asymmetric waveform to symmetric one by time-frequency conversion. As a result, we can obtain an average spectrum distortion ratio with 6.38% according to the pitch alteration ratio.

• Journal of Biomedical Engineering Research
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• v.13 no.4
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• pp.323-330
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• 1992

A Study of Image Compression Algorithm using DCT This paper describes the system that implement a JPEG(Joint Photographic Experts Group) algorithm based on DCT(Discrete Cosine Transform) uslng CCD kameva, Image Grabber, and IBM PC. After cosine transforms the acquisited image, this algorithm quantize and entropy encode the coefficients by JPEG code table. The coefficients are reconstructed by the Huffman decoding, dequantized procedure, and Inverse cosine transform. The results obtained from the impleulented system are as follows. (1) For effcient storage and easy implementation, this system save Image as a PCX formal (2) Thls system get 7:1 compression ratio(3.8 RMSE value) without large distortion. (3) With a low pass filtering, thls system eliminate high frequency components and get 20% enhanced compression ratio. (4) Thls system enhance the reconstructed Image using histogram modeling.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.2A
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• pp.187-193
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• 2006

Multicarrier systems has major shortcomings due to the large numbers of subcarrier. One of the shortcomings is the high peak-to-average power ratio(PAPR), which causes nonlinearity distortion in the transmitter, and degrades the performance of the system significantly. Partial transmit sequence(PTS) is one of the best approaches that can significantly improve the statistics of the PAPR of an orthogonal frequency division multiplexing(OFDM) signal. Guided scrambling(GS) is an extension of self-synchronizing scrambling. It is also capable of guiding the scrambling process to produce a balanced encoded bit stream. In this paper, we propose a sub-optimal algorithm to optimize the phase factor in PTS. In addition, we integrate GS coding with this method. The proposed Hybrid algorithm can get much more PAPR reduction at the expense of incur a few error extension and do not require transmission of side information.