• The Journal of the Acoustical Society of Korea
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• v.28 no.4
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• pp.392-400
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• 2009

In this paper, we propose an effective speech reinforcement technique base on ITU-T G.729A CS-ACELP codec under the near-end background noise environments. In general, since the intelligibility of the far-end speech for the near-end listener is significantly reduced under near-end noise environments, we require a far-end speech reinforcement approach to avoid this phenomena. In contrast to the conventional speech reinforcement algorithm, we reinforce the excitation signal of the codec's parameters received from the far-end speech signal based on the G.729A speech codec under various background noise environments. Specifically, we first estimate the excitation signal of ambient noise at the near-end through the encoder of the G.729A speech codec, reinforcing the excitation signal of the far-end speech transmitted from the far-end. we specially propose a novel approach to directly reinforce the excitation signal of far-end speech signal based on the decoder of the G.729A. The performance of the proposed algorithm is evaluated by the CCR (Comparison Category Rating) test of the method for subjective determination of transmission quality in ITU-T P.800 under various noise environments and shows better performances compared with conventional SNR Recovery methods.

• ETRI Journal
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• v.45 no.4
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• pp.615-626
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• 2023

Under limited transmission conditions, many factors affect the efficiency of video transmission. During the flight of an unmanned aerial vehicle (UAV), frequent network switching often occurs, and the channel transmission condition changes rapidly, resulting in low-video transmission efficiency. This paper presents an efficient video coding flow for UAVs working in the 5G nonstandalone network and proposes two bit controllers, including time and spatial bit controllers, in the flow. When the environment fluctuates significantly, the time bit controller adjusts the depth of the recursive codec to reduce the error propagation caused by excessive network inference. The spatial bit controller combines the spatial bit mask with the channel quality multiplier to adjust the bit allocation in space to allocate resources better and improve the efficiency of information carrying. In the spatial bit controller, a flexible mini graph is proposed to compute the channel quality multiplier. In this study, two bit controllers with end-to-end codec were combined, thereby constructing an efficient video coding flow. Many experiments have been performed in various environments. Concerning the multi-scale structural similarity index and peak signal-to-noise ratio, the performance of the coding flow is close to that of H.265 in the low bits per pixel area. With an increase in bits per pixel, the saturation bottleneck of the coding flow is at the same level as that of H.264.

• Journal of the Korea Society for Simulation
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• v.10 no.2
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• pp.15-24
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• 2001

In this paper we study the delay patterns of voice traffic for end-to-end users Caused by serving the whole bunch of applications traffic at the same time on the Voice over Internet Protocol (VoIP) network. Given the current situation that voice traffic is served along with other application services on the VoIP network, it is quite necessary to figure out how and by what the voice traffic requiring high QoS is delayed. We compare the delay performance of voice traffic on the VoIP network under FIFO with the one under Weighted Fair Queuing(WFQ), and discover the differences of the delay performance resulting from the use of different voice codec algorithms. The results of our study show that using the voice codec algorithm with a higher coding rate nd the queuing algorithm of WEQ can provide users with high-quality voice traffic.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.33 no.10B
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• pp.939-945
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• 2008

Tn this paper we described the process and the results of an implementation of Internet terminal using G.729.1 wideband speech codec for next generation network. For this purpose firstly we chose a high performance RISC application processor having DSP features for speech codec processing and enhanced Multimedia Accelerator(eMMA) function for video codec. In the implementation of this terminal, we used G.729.1 codec recently standardized in ITU-T which is a new scalable speech and audio codec that extends 0.729 speech coding standard. To adopt G.729.1 codec to this terminal we transformed most of the fixed point C codes which require more complexity into assembly codes so as to minimize processing time in the processor. As a result of this work we reduced the execution time of the original C codes about 80% and operated in real time on the terminal. For video we used H.263/MPEG-4 codec which is supported by the eMMA with hardware in the processor. In the SIP call processing test connected to real network we obtained under looms end-to-end delay and 3.8 MOS value measured with PESQ instrument. Besides this terminal operated well with commercial terminals.

• 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.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.9 no.8
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• pp.1696-1702
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• 2005

Real Time Transport Protocol (RTP) is the Internet standard protocol for transport of real time data audio/video IP Telephony, Multimedia Seivece. In case of 8kbps voice codec, the size of packet per data is 20bytes and become more large to minimal 40bytes with adding each layer's header in RTP/UDP/IP. To solve this problem, various header compression skill were suggested on point-to-point networks. But it compress even IP header and cannot be suitable to apply to end-to-end network Thus, We will renew header compression protocol to apply wired router-based network.

• Journal of Internet Computing and Services
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• v.11 no.4
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• pp.23-32
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• 2010

In this paper, we first propose the security enhancing of authentication protocol for Hash based RFID tag, and then a digital Codec for RFID tag is designed based on the proposed authentication protocol. The protocol is based on a three-way challenge response authentication protocol between the tags and a back-end server. In order to realize a secure cryptographic authentication mechanism, we modify three types of the protocol packets which defined in the ISO/IEC 18000-3 standard. Thus active attacks such as the Man-in-the-middle and Replay attacks can be easily protected. In order to verify effectiveness of the proposed protocol, a digital Codec for RFID tag is designed using Verilog HDL, and also synthesized using Synopsys Design Compiler with Hynix $0.25\;{\mu}m$ standard-cell library. Through security analysis and comparison result, we will show that the proposed scheme has better performance in user data confidentiality, tag anonymity, Man-in-the-middle attack prevention, replay attack, forgery resistance and location tracking.

• Journal of Korea Society of Digital Industry and Information Management
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• v.4 no.1
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• pp.41-46
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• 2008

With widespread use of the Internet and improvements in streaming media and compression technology, digital music, video, and image can be distributed instantaneously across the Internet to end-users. However, most conventional Digital Right Management are often not secure and fast enough to process the vast amount of data generated by the multimedia applications to meet the real-time constraints. In this paper, we propose the copyright protection using encryption of DCT coefficients and motion vector in MPEG-4 video codec of mobile device. This paper presents a new Digital Rights Management that modifies the Motion Vector of Macroblock for mobile device. Experimental results indicate that the proposed DRM can not only achieve very low cost of the encryption but also enable separable authentication to individual mobile devices such as Portable Multimedia Player and Personal Digital Assistants. The performance of the proposed methods have low complexity and low increase of bit rate in overhead.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.11 no.8
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• pp.1511-1518
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• 2007

As studies on ubiquitous computing are actively conducted, desire for various services, including image transmission storage, search and remote monitoring. has been expanding into mobile environment as well as to PCs. while CCTV (closed circuit TV) and un DVR (Digital video Recording) are used in places where security service such as intrusion detection system is required, these are high-end equipment. So it is not easy for ordinary users or household and small-sized companies to use them. Besides, they are difficult to be carried and camera solution for mobile device does not support high-quality function and provides low-definition of QVGA for picture quality. Therefore, in this study, design and implementation of embedded system of high-definition image transmission for ubiquitous mobile device which is not inferior to PC or DVR are described. To this end, usage of dedicated CPU for mobile device and design and implementation of MPEG-4 H/W CODEC also are examined. The implemented system showed excellent performance in mobile environment, in terms of speed, picture quality.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.11 no.8
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• pp.3076-3082
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• 2010

The techniques and the products which use various video compression codec are come out from army or civil field. In existing high-end PC environment, process of the video compression codec does not become a problem, but in embedded system environments which limited system resources, because the system load due to the high-resolution images compressed by high-density, issues of performance and utilization are highlighted. This paper proposes the DirectShow Filter interfaces which are a hardware method in order to solve the problem existing software algorithms for image compression performance and peripheral interfaces.