• 대한음성학회:학술대회논문집
• /
• 대한음성학회 2003년도 10월 학술대회지
• /
• pp.109-112
• /
• 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.

• 대한전자공학회:학술대회논문집
• /
• 대한전자공학회 1999년도 하계종합학술대회 논문집
• /
• pp.102-105
• /
• 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.

• 대한전자공학회:학술대회논문집
• /
• 대한전자공학회 2003년도 하계종합학술대회 논문집 Ⅳ
• /
• pp.2232-2235
• /
• 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.

• 한국방송∙미디어공학회:학술대회논문집
• /
• 한국방송공학회 2009년도 IWAIT
• /
• pp.326-329
• /
• 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)

• 한국방송∙미디어공학회:학술대회논문집
• /
• 한국방송공학회 1997년도 Proceedings International Workshop on New Video Media Technology
• /
• pp.37-42
• /
• 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.

• 한국정보과학회:학술대회논문집
• /
• 한국정보과학회 2006년도 가을 학술발표논문집 Vol.33 No.2 (D)
• /
• pp.512-517
• /
• 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
• /
• 제2권3호
• /
• pp.103-116
• /
• 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.

• 음성과학
• /
• 제11권4호
• /
• pp.67-73
• /
• 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.

• 한국음향학회지
• /
• 제28권7호
• /
• pp.618-629
• /
• 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.

• 전기전자학회논문지
• /
• 제13권4호
• /
• pp.43-46
• /
• 2009

음성신호는 화자에 대한 고유한 정보와 주변의 음향환경에 대한 정보는 물론 감정과 피로도 등 다양한 정보가 포함되어 있다. 이에 음성신호를 이용한 연구분야에서 감정 상태를 파악하기 위한 연구가 지속되어 왔다. 이에 본 논문에서는 화자의 감정을 인식하기 위해 ETSI의 3GPP2 표준코덱인 Selectable Mode Vocoder(SMV)를 분석한다. 이를 기반으로 감정 인식에 효과적인 특징들을 제안한다. 이후 선정된 특징 벡터를 이용하여 Gaussian Mixture Model(GMM) 기반의 감정 인식 알고리즘을 개발하고 Mixture component 개수를 변화시키면서 성능을 검증한다.