• Proceedings of the KSPS conference
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• 2003.10a
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• pp.109-112
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• 2003

In real time packetized voice applications, missing frames is a major source of voice quality degradation. Thus packet loss concealment(PLC) algorithms are needed to guarantee the QoS of the VoIP. Still current speech codecs for VoIP work poor when consecutive packet losses are issued. In this paper, we proposed a new PLC algorithm for the G.729 codec. Our algorithm works better especially when the consecutive packet loss occurs mainly because it adopts an adaptive gain controller utilizing the number of missing packet information combined with a fixed codebook vector estimation algorithm and LPC bandwidth expansion.

• Proceedings of the IEEK Conference
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• 1999.06a
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• pp.102-105
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• 1999

AAL type 2 provides for the bandwidth efficient transmission of low bit rate, short and variable packets in delay sensitive application. The service object for these networks ranges from POTS to multimedia conference. In this paper, we present one possible architecture which common part sublayer for new AAL type 2. The proposed CPS function has been achieved with on a FPGA The proposed architecture is faithful to the standardization of ITU-T and ATM-forum recommendation The proposed architecture applies to variable packet length from architecture CODECs for cellular network.. It's maximum process capability is 155Mbps with 256 CIDs. The architecture has sync./async. interface to application block and UTOPIA interface is used for physical layer interface.

• Proceedings of the IEEK Conference
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• 2003.07e
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• pp.2232-2235
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• 2003

In speech communication systems where two different speech codecs are interoperated, transcoding algorithm is a good approach because of its low complexity and improved synthesized speech quality. This paper proposes an efficient method to further improve the performance of transcoding algorithms as well as to reduce the complexity. In the conventional transcoding algorithms. a post-filter and a perceptual weighting filter should be operated sequentially because both decoding and encoding processes are needed. This results in the redundancy of the processing in terms of complexity and perceptual quality. Using the fact that their filter structures are similar, we replaced the two filters with one. The proposed algorithm requires 72.8% lower complexity than the conventional transcoding algorithm when we compare only the complexity of the filtering processes. The results of both objective and subjective tests verify that the proposed algorithm has slightly better quality than the conventional one.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2009.01a
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• pp.326-329
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• 2009

This paper proposes a new video compression algorithm using an adaptive transform that is adjusted depending on the frequency contents of the input signals. The adaptive transform is based on the warped discrete cosine transform (WDCT) which is shown to provide better performance than the DCT at high bit rates, when applied to JPEG compression scheme [1, 2, 3]. The WDCT is applied to the video compression in this paper, as a new feature in the H.264/AVC. The proposed method shows the coding gain over the H.264/AVC at high bit rates. The coding gain is shown over the 35dB PSNR quality, and the gain increases as the bit rate increases. (about 1.0dB at 45dB PSNR quality at maximum)

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 1997.06a
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• pp.37-42
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• 1997

This paper describes two different S/W-based video codec systems. One is a frame-based video codec with fixed structure based on ITU H.263 standard and the other an object-based video codec with flexible architecture based on ISO MPEG-4 standard currently under specification and planned to be finalized in 1998. These codes are an experimental implementations for examining the feasibility of real-time and/or flexible S/W-based video codecs operating in intra and/or internetworking environments.

• Proceedings of the Korean Information Science Society Conference
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• 2006.10d
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• pp.512-517
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• 2006

차세대 통신망은 IP 기반 핵심망을 중심으로 무선 액세스망을 비롯한 다양한 액세스망들이 접속되는 형태가 될 것으로 전망되고 있다. 이러한 망 환경에서 사용자가 만족할 수 있는 수준까지 멀티미디어 서비스를 제공하기 위해서는 서비스품질(QoS)이 필수적으로 보장되어야 한다. 특히, 종단간(end-to-end) QoS가 보장되어야 하는데, 이러한 종단간 QoS를 보장하기 위해서는 네트워크 QoS 뿐만 아니라 애플리케이션 계층에서의 QoS도 함께 지원되어야 한다. 본 논문에서는 네트워크에서 혼잡현상이 발생할 경우, 종단간 QoS를 제공하기 위해 송신측에서 가변 멀티코덱(애플리케이션 계층)을 사용하여 트래픽손실 및 지연을 줄일 수 있는 방법을 제안하고, 시뮬레이션을 이용한 성능 분석결과를 제시함으로써 제안된 방식의 타당성을 보인다.

• IEIE Transactions on Smart Processing and Computing
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• v.2 no.3
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• pp.103-116
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• 2013

This paper reviews various bitstream syntax and parser description (BSPD) languages used in MPEG multimedia standards. Traditionally, the bitstream syntax and semantics have been described in human-readable form. Several languages have been developed to describe the bitstream syntax using a computer-readable language and allow the automatic generation of bitstream parsing function from the description. The languages were designed for different objectives and applications but have a range of commonalities in functionality. The aim of this paper is to provide a historical overview of BSPD languages. The background and target application of the BSPD languages are reviewed. In addition, the technical features of each languages, including the linguistic basis (e.g., XML-based) and parser generation method, are discussed and evaluated. In addition, previous studies based on each language are introduced and categorized according to their objectives. Finally, the relevant technical issues that suggest the direction of the future researches are reported.

• Speech Sciences
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• v.11 no.4
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• pp.67-73
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• 2004

Packet loss, jitter, and delay in the Internet are caused mainly by the shortage of network bandwidth. It is due to queuing and routing process in the intermediate nodes of the packet network. In the Internet whose bandwidth is changing very rapidly in time depending on the number of users and data traffic, controlling the peak transmission bit-rate of a VoIP. system depending on the channel condition could be very helpful for making use of the available network bandwidth. Adapting packet size to the channel condition can reduce packet loss to improve the speech quality. It has been shown in [1] that a VoIP system with an AMR speech codec provides better speech quality than VoIP systems with fixed rate speech codecs. With the adaptive codec mode assignment. algorithm proposed in [1], in this paper, we performed the voice transmission experiments using the wireless LAN through the real Internet environment. Experimental results are analyzed and discussed with our findings.

• The Journal of the Acoustical Society of Korea
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• v.28 no.7
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• pp.618-629
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• 2009

The MPEG-4 Audio Lossless Coding (ALS) standard belongs to the family MPEG-4 audio coding standards. In contrast to lossy codecs such as AAC, which merely strive to preserve the subjective audio quality, lossless coding preserves every single bit of the original audio data. The ALS core codec is based on forward-adaptive linear prediction, which combines remarkable compression with low complexity. Additional features include long-term prediction, multichannel coding, and compression of floating-point audio material. This paper describes the basic elements of the ALS codec with a focus on prediction, entropy coding, and related tools and points out the most important applications of this standardized lossless audio format.

• Journal of IKEEE
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• v.13 no.4
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• pp.43-46
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• 2009

This paper deals with feeling recognition from people's voice to fine feature vectors. Voice signals include the people's own information and but also people's feelings and fatigues. So, many researches are being progressed to fine the feelings from people's voice. In this paper, We analysis Selectable Mode Vocoder(SMV) that is one of the standard 3GPP2 codecs of ETSI. From the analyzed result, we propose voices features for recognizing feelings. And then, feeling recognition algorithm based on gaussian mixture model(GMM) is proposed. It uses feature vectors is suggested. We verify the performance of this algorithm from changing the mixture component.