• The Journal of the Korea Contents Association
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• v.7 no.4
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• pp.192-198
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• 2007

Due to the development of information telecommunication technology, e-learning industry is rapidly expanding its scope along with its production technology. The recent trend of e-learning program is likely converted from Wmv(Window Media Video) of Microsoft to Flv(Flash video), which has less capacity but better quality than other image file. It has successfully drawn the users attention since Flv can operate at most OS environments and browsers let alone with window and Lenux without extra players and codec setup. However, there is no accurate data on comparative analysis between Wmv and Flv regarding capacity, quality and production time. Therefore, the study shows the comparative data analysis on Wmv and Flv so as to set out production platform up to its idiosyncrasy.

• The Journal of the Korea Contents Association
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• v.9 no.11
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• pp.10-17
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• 2009

Wyner-Ziv video coding is a new video compression paradigm based on distributed source coding theory of Slepian-Wolf and Wyner-Ziv. Wyner-Ziv coding enables light-encoder/heavy-decoder structure by shifting complex modules including motion estimation/compensation task to the decoder. Instead of performing the complicated motion estimation process in the encoder, the Wyner-Ziv decoder performs the motion estimation for the generation of side information in order to make the predicted signal of the Wyner-Ziv frame. The efficiency of side information generation deeply affects the overall coding performance, since the bit-rates of the Wyner-Ziv coding is directly dependent on side information. In this paper, an improved side information generation method using field coding is proposed. In the proposed method, top fields are coded with the existing SI generation method and bottom fields are coded with new SI generation method using the information of the top fields. Simulation results show that the proposed method improves the quality of the side information and rate-distortion performance compared to the conventional method.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.14 no.9
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• pp.1957-1964
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• 2010

In Video coding, the first frame is encoded in intra mode which generates a larger number of bits. In addition, the first frame is used for the inter mode encoding of the following frames. Thus the intial QP for the first frame of GOP affects the first frame as well as the following frames. Traditionally, the initial QP of a GOP is determined by the initial QP of the previous GOP and the average QP of the inter mode frames. In case of JM, the initial QP of a GOP is adjusted as the initial QP being less than the average QP of inter mode frames by two. However, this method is not suitable for the low bit rate video coding. In this paper, the linear relationship between the optimal QP and the ratio of the PSNR of the first frame and the average PSNR of the inter mode frames is first investigated and the linear model is proposed based on the results of the investigation. The proposed model calculate the optimal initial QP using the encoding results of the previous GOP. It is shown by experimental results that the new algorithm can predict the optimal initial QP more accurately and generate the PSNR performance better than that of the existing JM algorithm.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.42 no.3 s.303
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• pp.31-40
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• 2005

In this paper, we develop an approach toward JSC(Joint Source-Channel Coding) method for MPEG-4 based FGS(Fine Granular Scalability) video coding and transmission in fixed and mobile receiving environment(Digital Audio Broadcasting, DAB). The source coder used MPEG-4 FGS video codec, the channel coder used RCPC(Rate Compatible Punctured Convolution) code and the modulation method used QPSK modulation. We have considered channel environment of AWGN and mobile receiving environment. This study determined optimum Trade-off point between source bit rate and channel coding rate in variable channel states. We compared FGS-JSC method and general single layer CBR(Constant Bit Rate) transmission. In this results, FGS-JSC was appeared better performance than CBR transmission.

• The Journal of the Acoustical Society of Korea
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• v.21 no.3
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• pp.304-309
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• 2002

We propose the computation reduction method of real root method that is mainly used in the CELP (Code Excited Linear Prediction) vocoder. The real root method is that if polynomial equations have the real roots, we are able to find those and transform them into LSP. However, this method takes much time to compute, because the root searching is processed sequentially in frequency region. In this paper, to reduce the computation time of real root, we compare the real root method with two methods. In first method, we use the mal scale of searching frequency region that is linear below 1 kHz and logarithmic above. In second method, The searching frequency region and searching interval are ordered by each coefficient's distribution. In order to compare real root method with proposed methods, we measured the following two. First, we compared the position of transformed LSP (Line Spectrum Pairs) parameters in the proposed methods with these of real root method. Second, we measured how long computation time is reduced. The experimental results of both methods that the searching time was reduced by about 47% in average without the change of LSP parameters.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.48 no.5
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• pp.102-108
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• 2011

Support vector machines are well known for their outstanding performance in pattern recognition fields. One example of their applications is music/speech classification for a standardized codec such as 3GPP2 selectable mode vocoder. In this paper, we propose a novel scheme that improves the speech/music classification of support vector machines based on the second-order conditional maximum a priori. While conventional support vector machine optimization techniques apply during training phase, the proposed technique can be adopted in classification phase. In this regard, the proposed approach can be developed and employed in parallel with conventional optimizations, resulting in synergistic boost in classification performance. According to experimental results, the proposed algorithm shows its compatibility and potential for improving the performance of support vector machines.

• Journal of the Institute of Electronics and Information Engineers
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• v.50 no.1
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• pp.86-96
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• 2013

Unified Speech and Audio Coding (USAC) is the speech/audio codec with the best quality, approved on Final Draft International Standard (FDIS) at MPEG meeting in 2011. Since MPEG conventionally standardizes only the decoder, it is not easy to study on the encoder technologies. Furthermore, Reference Model(RM) shows extremely poor performance. To solve these problems, the open source project(JAME) proposes the methods to make the improved performance of main encoder technologies in USAC. Especially, this paper introduces the encoder modules: the signal classifier for selective operation between two coders, the psychoacoustic model in frequency domain, and window transition technology. Finally, the results of verification test for FDIS and the performance of Common Encoder are appended.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.14 no.1
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• pp.197-204
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• 2010

This paper proposes a new body-effect compensated switch configuration for low voltage and low distortion switched-capacitor (SC) applications. The proposed circuit allows rail-to-rail switching operation for low voltage SC circuits and has better total harmonic distortion than the conventional bootstrapped circuit by 19 dB. A 2-1 cascaded sigma-delta modulator is provided for performing the high-resolution analog-to-digital conversion on audio codec in a communication transceiver. An experimental prototype for a single-stage folded-cascode operational amplifier (opamp) and a 2-1 cascaded sigma-delta modulator has been implemented m a 0.25 micron double-poly, triple-metal standard CMOS process with 2.7 V of supply voltage. The 1% settling time of the opamp is measured to be 560 ns with load capacitance of 16 pF. The experimental testing of the sigma-delta modulator with bit-stream inspection and analog spectrum analyzing plot is performed. The die size is $1.9{\times}1.5\;mm$.

• The KIPS Transactions:PartC
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• v.9C no.3
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• pp.375-384
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• 2002

In the current Internet, the QoS of interactive applications is hardly guaranteed because of variable bandwidth, packet loss and delay. Moreover, VoIP which is becoming an important part of the information infra-structure in these days, is susceptible to network packet loss and end-to-end delay. Therefore, it needs error control mechanisms in network level or application level. The FEC-based error control mechanisms are used for interactive audio application such as VoIP. The FEC sends a main information along with redundant information to recover the lost packets and adjusts redundant information depending on network conditions to reduce the bandwidth overhead. However, because most of the error control mechanisms do not consider end-to-end delay but packet loss rate, their performances are poor. In this paper, we propose a new error control algorithm, SCCRP, considering packet loss rate as well as end-to-end delay. Through experiments, we confirm that the SCCRP has a lower packet loss rate and a lower end-to-end delay after reconstruction.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.1C
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• pp.72-82
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• 2004

This paper proposes more robust error concealment techniques (ECTs) for MPEG-2 intra coded frame. MPEG-2 source coding algorithm is very sensitive to transmission errors due to the use of variable-length coding. The transmission errors are corrected by error correction scheme, however, they cannot be revised properly. Error concealment (EC) is used to conceal the errors which are not corrected by error correction and to provide minimum visual distortion at the decoder. If errors are generated in intra coded frame, that is the starting frame of GOP, they are propagated to other inter coded frames due to the nature of motion compensated prediction coding. Such propagation of error may cause severe visual distortion. The proposed algorithm in this paper utilizes the spatio-temporal information of neighboring inter coded frames to conceal the successive slices errors occurred in I-frame. The proposed method also overcomes the problems that previous ECTs reside. The proposed algorithm generates consistent performance even in network where the violent transmission errors frequently occur. Algorithm is performed in MPEG-2 video codec and we can confirm that the proposed algorithm provides less visible distortion and higher PSNR than other approaches through simulations.