• Journal of the Institute of Electronics Engineers of Korea SP
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• v.41 no.5
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• pp.203-216
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• 2004

Recent advances in video coding technology have resulted in rapid growth of application in mobile communication, With this explosive growth reliable transmission and error resilient technique become increasingly necessary to offer high quality multimedia service. In this paper, we present the result of our investigation on the error resilient performance evaluation of the MPEG-4 simple profile under the H.324/M and the H.264/AVC baseline under the IP packet networks. Especially, we have tested error resilient tools of MPEG-4 simple profile such as resynchronization marker insertion, data partitioning, and of H.264/AVC baseline such as the flexible macroblock ordering (FMO) scheme. The objective quality of decoded video is measured in terms of rate and PSNR under various random bit and burst error conditions.

• The Journal of the Acoustical Society of Korea
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• v.26 no.3
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• pp.109-114
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• 2007

In this paper, we Propose a pyramid VQ to quantize the transform coefficients of TCX module for the audio improvement of AMR-WB+ codec. The Proposed pyramid VQ is compared to the $RE_8$ Lattice VQ used in the AMR-WB+ standard codec. demonstrating improvement 4% and 5.7%. respectively, in Mean Squared Error (MSE) and 3.3% and 4.7%. respectively, in Perceptual Evaluation of Audio Quality (PEAQ) by 8-dimensional and 16-dimensional Pyramid VQ.

• Journal of Internet Computing and Services
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• v.18 no.5
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• pp.17-22
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• 2017

In a MPC coding using excitation source of voiced and unvoiced, it would be a distortion of speech waveform. This is caused by normalization of synthesis speech waveform of voiced in the process of restoration the multi-pulses of representation section. This paper present PCFBD-MPC( Position Compensation Frequency Band Division-Multi Pulse Coding ) used V/UV/S( Voiced / Unvoiced / Silence ) switching, position compensation in a multi-pulses each pitch interval and Unvoiced approximate-synthesis by using specific frequency in order to reduce distortion of synthesis waveform. Also, I was implemented that the PCFBD-MPC( Position Compensation Frequency Band Division-Multi Pulse Coding ) system and evaluate the SNRseg of PCFBD-MPC in coding condition of 8kbps. As a result, SNRseg of PCFBD-MPC was 13.4dB for female voice and 13.8dB for male voice respectively. In the future, I will study the evaluation of the sound quality of 8kbps speech coding method that simultaneously compensation the amplitude and position of multi-pulse source. These methods are expected to be applied to a method of speech coding using sound source in a low bit rate such as a cellular phone or a smart phone.

• Journal of Computing Science and Engineering
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• v.7 no.3
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• pp.168-176
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• 2013

Long-term evolution (LTE) is emerging as a major candidate for 4G cellular networks to satisfy the increasing demands for mobile broadband services, particularly multimedia delivery. Multiple-input multiple-output (MIMO) technology combined with orthogonal frequency division multiple access and more efficient modulation/coding schemes (MCS) are key physical layer technologies in LTE networks. However, in order to fully utilize the benefits of the advances in physical layer technologies, the MIMO configuration and MCS need to be dynamically adjusted to derive the promised gains of 4G at the application level. This paper provides a performance evaluation of video traffic with variations in the physical layer transmission parameters to suit the varying channel conditions. A quantitative analysis is provided using the perceived video quality as a video quality measure (evaluated using no-reference blocking and blurring metrics), as well as transmission delay. Experiments are performed to measure the performance with changes in modulation and code rates in poor and good channel conditions. We discuss how an adaptive scheme can optimize the performance over a varying channel.

• The Journal of the Acoustical Society of Korea
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• v.15 no.2
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• pp.55-60
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• 1996

This paper describes the subjective assessment of the multi-channel audio quality according to the bitrate changing and evaluates the predistortion effect to avoid the unmasked noise after matrixing/dematrxing process in transmission and regeneration of the multi-channel audio. The simulation is processed by the perceptual coding that is MPEG-2 Audio layer II algorithm. We evaluate the quality improvement about predistortion using or not by 384, 320, 256, 128kbps. As the result of the double blind subjective assessment, 5 Grade-Impairment Scale is scored under minus one to 320kbps and so audio quality is evaluated to be perceptible, but not annoying in 3/2 channel. The effect of the predistortion is improved one level in 128kbps and especially speech test material I better improved than music test materials.

• IE interfaces
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• v.19 no.3
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• pp.236-244
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• 2006

Postal code is essential for improving efficiency of mail sorting. The Korean postal code, which is introduced in 1970 and had been reformed two times, has controversial issues because of changed postal environments such as modification of administrative districts and logistical transformation into a hub-and-spokes structure. In this study, we propose several improvement and evaluation methods in order to prepare an alternative plan of postal code system. After performing status analysis, we define the customers of postal code and analyze the requirements of them. The 6 proposed new postal code systems are based on each customer groups’ requirements. To evaluate the new systems, we develop several quantitative and qualitative indexes and make a survey of postal experts. Besides, we evaluate the influence of them to the postal environments including delivery sequence sorting process.

• Journal of Digital Contents Society
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• v.16 no.4
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• pp.583-593
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• 2015

Today, caused by the growth of computer and communication technology, multimedia, especially image data are being used in different application divisions. JPEG2000 that is widely used these days provides a Region-of-Interest(ROI) technique. The extraction of ROI has to be rapidly executed and automatically extracted in a huge amount of image because of being seen preferentially to the users. For this purpose, this paper proposes a method about preferential processing and automatic extraction of ROI using the distribution of edge in the code block of JPEG2000. The steps are the extracting edges, automatical extracting of a practical ROI, grouping the ROI using the ROI blocks, generating the mask blocks and then quantization, ROI coding which is the preferential processing, and EBCOT. In this paper, to show usefulness of the method, we experiment its performance using other methods, and executes the quality evaluation with PSNR between the images not coding an ROI and coding it.

• The Journal of the Korea institute of electronic communication sciences
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• v.7 no.2
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• pp.243-248
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• 2012

In this paper we simulate the capacity of VOIP calls through WiFi network by computer simulations using OPNET modeler. The results show that sudden quality degradations occur on all VoIP calls when the number of call of an AP(Access Point) increases beyond a specific value. The reason of the quality degradation was turned out to be the queueing delay at the down link of AP. Under the IEEE 802.11b environments, the maximum number of VoIP calls of an AP maintaining the required voice quality (MOS > 2.5), was evaluated as 5, 12, and 27 when we use G.711, G.729a, and G.729a VAD codec, respectively.

• The KIPS Transactions:PartB
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• v.15B no.5
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• pp.383-390
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• 2008

Block based transform coding is most popular approach for image and video compression. However it suffers from severe quality degradation especially from blocking artifacts. The subjective quality degradation caused by such blocking artifacts in general does not agree well with an objecive quality measurement such as PSNR. Hence new quality evaluation technique is necessary. We propose a new image quality assessment method by measuring blocking artifacts for block based transform coded images. In order to characterize blocking artifacts, proposed method utilizes the facts that, blocking artifacts, when occur, have different pixel values along the block boundaries and such differences usually continuously span along the whole boundaries. This method does not require the original uncompressed image. It operates on single block boundary and quantifies the amount of blocking artifacts on it. Experiments on various compressed images various bitrates show that proposed quantitative measure of blocking artifacts matches well with the subjective quality of them judged by human visual system.

• 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.