• Journal of the Korea Institute of Military Science and Technology
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• v.20 no.3
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• pp.431-440
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• 2017

HF communications are used as a last means of long distance communications without any relay node in NLOS (Non Line-Of-Sight) environment. Conventional analog voice communication in the HF band is vulnerable to security as well as severe background noise. To overcome these shortcoming, digital voice was introduced into HF radios in the early 1980s. In this paper, we analyze avaliable digital voice communication ground wave range of HF radios that applied MELPe CODEC and MIL-STD-188-110B physical layer standard using GRWAVE program. And we evaluate usefulness of digital voice communication in HF band.

• Journal of Internet Computing and Services
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• v.17 no.4
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• pp.11-17
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• 2016

In this paper, we proposed a multi-viewer system with multiple HD cameras based AVB(Audio Video Bridge) ethernet cable using IP networking, and FPGA(Xilinx Zynq 702) for bus safety systems. This AVB (IEEE802.1BA) system can be designed for the low latency based on FPGA, and transmit real-time with HD video and audio signals in a vehicle network. The proposed multi-viewer platform can multiplex H.264 video signals from 4 wide-angle HD cameras with existed ethernet 1Gbps. and 2-wire 100Mbps cables. The design of Zynq 702 based low latency to H.264 AVC CODEC was proposed for the minimization of time-delay in the HD video transmission of car area network, too. And the performance of PSNR(Peak Signal-to-noise-ratio) was analyzed with the reference model JM for encoding and decoding results in H.264 AVC CODEC. These PSNR values can be confirmed according the theoretical and HW result from the signal of H.264 AVC CODEC based on Zynq 702 the multi-viewer with multiple cameras. As a result, proposed AVB multi-viewer platform with multiple cameras can be used for the surveillance of audio and video around a bus for the safety due to the low latency of H.264 AVC CODEC design.

• The Transactions of the Korea Information Processing Society
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• v.3 no.4
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• pp.938-946
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• 1996

Hierarchical subband codec has been widely promoted in the field of data compression/decompression because of their simplicity and modular nature. Over the past years, the study has received great attention to the perfect reconstruction (PR)system which perfectly recovers the original input signal at the reconstructed output. However, in the actual subband codec system, the signals that passed through the analysis filter bank are quantized before transmission to the receiver side and reconstructed by the synthesis filter bank. Thus the PR system is impossible and the quantization effects must be carefully considered in the system design such that the system recovers the reconstructed output as possible to the the original input signal with minimum quantization error.In this paper, we propose an optimum hierarchical subband codec structure in the presence of quantizer. The optimality criteria of the code is given to the deign of the hierarchical analysis/synthesis subband filter bank and the quantizer that minimize then output mean square error due to the quantizer in the codec. Specific opti-mum design esamples are shown with level-1, level-2 hierarchical structure. The optimal designs are verified by computer simulation.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.39 no.5
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• pp.477-485
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• 2002

In this paper, we present an efficient hardware architecture for the H.263 video codec and its VLSI implementation. This architecture is based on the unified interface by which internal hardware engines and an internal RISC processor are connected one another. The unified interface enables the modular design of internal blocks, efficient hardware/software partitioning, and pipelined paralled operations. The developed VLSI supports the H.263 version 2 profile 3 @ level 10, and moreover, both the control protocol H.245 and the multiplexing protocol H.223. Therefore, it can be used for the complete ITU-T H.324 or 3GPP 3G 324M multimedia processor with the help of an external audio codec. Simultaneous encoding and decoding of QCIF format images at a rate greater than 15 frames per second is achieved at 40 MHz clock frequency.

• Proceedings of the IEEK Conference
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• 2000.09a
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• pp.827-830
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• 2000

3GPP and ETSI adopted AMR(Adaptive Multi-Rate) as a standard for next generation IMT-2000 service. In this paper, we analyzed algorithm about AMR and optimized ANSI C source on the C complier and assembly language of Texas Instrument . The implemented AMR speech codec requires 28.2MIPS of complexity for encoder and 5.5MIPS for decoder. we performed real-time implementation of AMR speech codec using 82% of TMS320C5402 with 40 MIPS specification. We give proof that the output speech of the implemented speech codec on DSP board is identical with result of C source program simulation. Also the reconstructed speech is verified in the real-time environment consisted of microphone and speaker.

• Journal of Broadcast Engineering
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• v.25 no.2
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• pp.233-241
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• 2020

In this paper, a new buffer control method is proposed for improving the performance of the frequency domain part of the 3D audio (3DA) core codec. For the proposed buffer control method, we first combine the 3DA RM9 with the 3GPP AAC buffer control method which includes the psychoacoustic model and rate-distortion control process with the spectral hole avoidance algorithm. Then, we revise the 3GPP buffer control method so as to achieve a faithful bit allocation to the frames with higher activity. With the MUSHRA test, we prove that the proposed buffer control method demonstrates better performance than the 3DA RM9 and 3GPP AAC.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.33 no.10B
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• pp.919-927
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• 2008

The growing availability of broadband multimedia applications are providing strict real-time services and enabling users to feel much interactivity as well as 'sense of presence'. In that sense, there would be increased demand for what network and system have to do to fulfill the user expectations. In this paper, we implement a real-time multimedia application which make use of real-time DXT CODEC and investigate what the challenges of networking and system are. We carry out various experiments over a research network test-bed to evaluate the performance of our proposals.

• Journal of the Korea Society of Computer and Information
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• v.14 no.8
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• pp.51-58
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• 2009

In this paper. we propose search algorithm using Low Peak Feature of audio and image color value by which similar movies can be identified. Combing through entire video files for the purpose of recognizing and retrieving matching movies requires much time and memory space. Moreover, these methods still share a critical problem of erroneously recognizing as being different matching videos that have been altered only in resolution or converted merely with a different codec. Thus we present here a similar-video-retrieval method that relies on analysis of audio patterns, whose peak features are not greatly affected by changes in the resolution or codec used and image color values. which are used for similarity comparison. The method showed a 97.7% search success rate, given a set of 2,000 video files whose audio-bit-rate had been altered or were purposefully written in a different codec.

• The Transactions of the Korean Institute of Electrical Engineers P
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• v.59 no.4
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• pp.390-394
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• 2010

This paper describes a study on developing an insert and detect system of digital watermark to the MPEG-2 coder and decoder. Digital watermark inserting system substitutes a coefficient among the DCT coefficients for new one. This replacement performs in MPEG-2 PS format. Watermark detecting system analyzes the input data stream of MPEG-2 and detect the probability of occurrence of a specific code or watermarking code. Our inserting system had very high invisibility and the PSNR of the results was 130 dB around. And our detection rate of watermark was over 99% in digital format and over 85% in analog captured format according to the quality of contents.

• Journal of Broadcast Engineering
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• v.2 no.1
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• pp.24-35
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• 1997

Basic and transmission video quality testing was performed in real-time mode using hardware video codec based on MPEG-2 MP@ML standard including subsequent transmitting and receiving satellite simulator unit. The double-stimulus Impairment scale method and the double-stimulus continous quality scale method based on CCIR 500-5 were used as an evaluation method. The whole digital broadcasting satellite system consisting of MPEG-2 codec, system mux/demux, channel codec, channel, modem, antenna, etc. was put into the overall video quality testbed and the basic and transmission error quality assessment was performed at various bitrates and BER for an integrated system performance evaluation. In transmission error video quality testing, transmission error video quality maintained on average above 3.9 point on the 5-point scale. The low-bit rate quality such as film mode(@2Mbps) highly depended on the statitical characteristics of video source and maintained on average around 2.7 point.