• The Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.14 no.4
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• pp.247-256
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• 2021

In this paper, we propose a packet loss concealment (PLC) algorithm using pitch harmonic motion prediction and adaptive signal amplitude prediction and. The spectral motion prediction method divides the spectral motion of the previous usable frame into predetermined sub-bands to predict and restore the motion of the lost signal. In the proposed algorithm, the speech signal is classified into voiced and unvoiced sounds. In the case of voiced sounds, it is further divided into pitch harmonics using the pitch frequency to predict and restore the pitch harmonic motion of the lost frame, and for the unvoiced sound, the lost frame is restored using the spectral motion prediction method. When the continuous loss of speech frames occurs, a method of adjusting the gain using the least mean square (LMS) predictor is proposed. The performance of the proposed algorithm was evaluated through the objective evaluation method, PESQ (Perceptual Evaluation of Speech Quality) and was showed MOS 0.1 improvement over the conventional method.

• The Transactions of The Korean Institute of Electrical Engineers
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• v.59 no.4
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• pp.831-840
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• 2010

In this paper, we propose a hearing impairment simulator considering reduced frequency selectivity and asymmetrical auditory filter of the hearing impaired, and we verified the reduced frequency selectivity and asymmetrical auditory filter affected in speech perception through experiments. The reduced frequency selectivity has made embodied by spectral smearing using LPC(linear prediction coding). The shapes of auditory filter are asymmetrical different with each center frequency. Hearing impaired person which has hearing loss was differently changed with that of normal hearing people and it has different value for speech of quality through auditory filter. The experiments confirmed subjective test and objective test. The subjective experiments are composed of 4 kinds of tests: pure tone test, SRT(speech reception threshold) test, and WRS(word recognition score) test without spectral smearing, and WRS test with spectral smearing. The experiment of the hearing impairment simulator was performed from 9 subjects who have normal ears. The amount of spectral smearing was controlled by LPC order. The asymmetrical auditory filter of proposed hearing impairment simulator was simulated and then some tests to estimate the filter's performance objectively were performed. The objective experiment as simulated auditory filter's performance evaluation method used PESQ(perceptual evaluation of speech quality) and LLR(log likelihood ratio) for speech through auditory filter. The processed speech was evaluated objective speech quality and distortion using PESQ and LLR value. When hearing loss processed, PESQ and LLR value have big difference according to asymmetrical auditory filter in hearing impairment simulator.

• The Journal of the Acoustical Society of Korea
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• v.21 no.6
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• pp.531-542
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• 2002

Novel tandemless transcoding method for AMR and EVRC speech coders is proposed in this paper. In contrast to conventional tandem method, the parameters which is used commonly in speech coder where CELP algorithm is adapted are directly transcoded. The proposed algorithm is composed of LSP transcoding, pitch delay transcoding, gains transcoding and fixed codebook vector transcoding Evaluation results show that the novel algorithm achieves better speech quality than tandem method and reduce computational complexity and delay.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.7 no.5
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• pp.1108-1130
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• 2013

Wireless Body Area Networks (WBANs) are becoming increasingly important to solve the issue of health care. IEEE 802.15.6 is a wireless communication standard for WBANs, aiming to provide a real-time and continuous monitoring. In this paper, we present our development of a modified Markov Chain model and a backoff model, in which most features such as user priorities, contention windows, modulation and coding schemes (MCSs), and frozen states are taken into account. Then we calculate the normalized throughput and average access delay of IEEE 802.15.6 networks under saturation and ideal channel conditions. We make an evaluation of network performances by comparing with IEEE 802.15.4 and the results validate that IEEE 802.15.6 networks can provide high quality of service (QoS) for nodes with high priorities.

• Proceedings of the KAIS Fall Conference
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• 2010.11a
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• pp.402-405
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• 2010

본 논문에서는 소스 코드에 자주 나타나는 코딩 패턴을 자동으로 추출하기 위하여 소스 코드를 분석하고 특성을 평가한다. 기존에 소스 코드에 대한 패턴 마이닝을 이용한 코딩 패턴 감지 방법이 제안되었지만, 수동으로 조사 가능한 코딩 패턴의 수는 한정되어 있기 때문에 대규모 소프트웨어 등에 대한 충분한 분석은 할 수 없었다. 따라서 본 논문에서는 개발자가 분석하고자 하는 코딩 패턴에 대한 자동 추출을 목표로 코딩 패턴의 특성 평가 지표를 선정하여 소스 코드에 대한 분석을 수행한다.

• Journal of the Korea Society of Computer and Information
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• v.3 no.3
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• pp.99-106
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• 1998

This Paper proposes a novel fractal image method for image communication to shorten time to take on fractal encoding by using limited search area method and scaling method First, the original image is contracted respectively by half and by quarter with the scaling method. And then, the corresponding domain block of the quarter-sized image which is most similar with one range block of the half-sized image is searched within the limited area in order to reduce the encoding time extremely As the result of the evaluation, the proposed algorithm provided much shorter encoding time and better compression ratio with a little degradation of the decoded image quality than Jacquin's method.

• ETRI Journal
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• v.34 no.3
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• pp.474-477
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• 2012

The MPEG-D unified speech and audio coding (USAC) standardization process was initiated by MPEG to develop an audio codec that is able to provide consistent quality for mixed speech and music contents. The current USAC reference model structure consists of frequency domain (FD) and linear prediction domain (LPD) core modules and is controlled using a signal classifier tool. In this letter, we propose an LPD single-mode USAC structure using an adaptive widowing-based transform-coded excitation module. We tested our system using official test items for all mono-evaluation modes. The results of the experiment show that the objective and subjective performances of the proposed single-mode USAC system are better than those of the FD/LPD dual-mode USAC system.

• Proceedings of the KSRS Conference
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• 2002.10a
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• pp.88-88
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• 2002

Satellite Image archiving system requires large storage and long transmission time. A simple and cheap way of overcoming these limitations is to increase the compression ratio. However this requires a feasibility study for accurate applications. Here, a new still image compression standard is being developed, the JPEG2000. It provides lossless and lossy compression, progressive transmission by pixel accuracy and by resolution, region-of-interest coding, user-defined tiling size, random codestream access and processing etc. In this study, we will briefly introduce the JPEG2000 compression standard which provides a new compression technique based on the wavelet technology and offers better compression ratios, and evaluate the compression ratios of JPEG2000 for satellite image by performing various image quality tests. Also, we will compare brief test result using the commercial remote sensing software.

• Journal of Broadcast Engineering
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• v.21 no.6
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• pp.929-943
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• 2016

This paper proposes the rate control algorithm based on the S-JND (Saliency-Just Noticeable Difference) model for considering perceptual visual quality. The proposed rate control algorithm employs the S-JND model to simultaneously reflect human visual sensitivity and human visual attention for considering characteristics of human visual system. During allocating bits for CTU (Coding Tree Unit) level in a rate control, the bit allocation model calculates the S-JND threshold of each CTU in a picture. The threshold of each CTU is used for adaptively allocating a proper number of bits; thus, the proposed bit allocation model can improve perceptual visual quality. For performance evaluation of the proposed algorithm, the proposed algorithm was implemented on HM 16.9 and tested for sequences in Class B and Class C under the CTC (Common Test Condition) RA (Random Access), Low-delay B and Low-delay P case. Experimental results show that the proposed method reduces the bit-rate of 2.3%, and improves BD-PSNR of 0.07dB and bit-rate accuracy of 0.06% on average. We achieved MOS improvement of 0.03 with the proposed method, compared with the conventional method based on DSCQS (Double Stimulus Continuous Quality Scale).

• The Transactions of The Korean Institute of Electrical Engineers
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• v.60 no.6
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• pp.1196-1203
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• 2011

The reduced frequency selectivity is a typical phenomenon of sensorineural hearing loss. In this paper, we compared two modeling methods for reduced frequency selectivity of hearing impaired people. The two models of reduced frequency selectivity were made using LPC(linear prediction coding) algorithm and bandwidth control algorithm based on ERB(equivalent rectangular bandwidth) of auditory filter, respectively. To compare the effectiveness of two models, we compared the result of PESQ (perceptual evaluation of speech quality) and LLR(log likelihood ratio) using 36 Korean words of two syllables. To verify the effect on noise condition, we mixed white and babble noise with 0dB and -3dB SNR to speech words. As the result, it is confirmed that the PESQ score of bandwidth control algorithm is higher than the score of LPC algorithm, on the other hands, and the LLR score of LPC algorithm is lower than the score of bandwidth control algorithm. It means that both non-linearity and widen auditory filter characteristics caused by reduced frequency selectivity could be more reflected in bandwidth control algorithm than in LPC algorithm.