• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.13 no.2
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• pp.176-187
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• 2002

The digital Audio Broadcasting (DAB) system adopts Coded OFDM(COFDM) for channel coding. The COFDM is a combined technique of multicarrier transmission(OFDM) and punctured convolutional coding with viterbi error correction. Because the channel coding is an important topic for OFDM systems, this paper proposes a new turbo coded OFDM system that replaces the existing RCPC codec by a turbo codec without modifying the puncturing procedure and puncturing vectors defined in the standard DAB system for compatibility. The performance of a new system is compared to that of the conventional system under the frequency selective Rician fading channel and the frequency selective Rayleigh fading channel in conjunction with DAB transmission mode I suitable for the terrestrial single frequency network(SFN) broadcasting. The standard system's performance was improved with the aid of turbo codec.

• ETRI Journal
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• v.25 no.6
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• pp.489-502
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• 2003

This paper presents an MPEG-4 video codec, called MoVa, for video coding applications that adopts 3G-324M. We designed MoVa to be optimal by embedding a cost-effective ARM7TDMI core and partitioning it into hardwired blocks and firmware blocks to provide a reasonable tradeoff between computational requirements, power consumption, and programmability. Typical hardwired blocks are motion estimation and motion compensation, discrete cosine transform and quantization, and variable length coding and decoding, while intra refresh, rate control, error resilience, error concealment, etc. are implemented by software. MoVa has a pipeline structure and its operation is performed in four stages at encoding and in three stages at decoding. It meets the requirements of MPEG-4 SP@L2 and can perform either 30 frames/s (fps) of QCIF or SQCIF, or 7.5 fps (in codec mode) to 15 fps (in encode/decode mode) of CIF at a maximum clock rate of 27 MHz for 128 kbps or 144 kbps. MoVa can be applied to many video systems requiring a high bit rate and various video formats, such as videophone, videoconferencing, surveillance, news, and entertainment.

• 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.

• Journal of Korea Multimedia Society
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• v.4 no.1
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• pp.75-81
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• 2001

A EBCOT(Embedded Block Coding with Optimized Truncation) is image compression codec using in JPEG2000, currently the new standard for still image coding. this paper proposes multi-channel video transfer system with EBCOT using a single codec to transfer video to difference band-width channel. This parer testify that compression rate of EBCOT higher than ordinary VLC using RLC and Huffman codec and apply EBCOT to JPEG structure. this structure increases parallelism and error resilience using black coding method. finally it looks into difficult to apply MPEG structure to multi channel video transfer system, and proposes solution using EBCOT.

• Journal of Broadcast Engineering
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• v.10 no.4 s.29
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• pp.515-527
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• 2005

A next-generation codec should be developed to be a scalable video codec(SVC) that not only maximizes the coding efficiency but also adaptively copes with the various communication devices and the variation of network environments. To meet these requirements, Joint Video Team (JVT) of ISO/IEC and ITU-T is standardizing H.264/AVC based SVC. In this paper, we introduce research directions and status on SVC standardization and also analyze techniques and algorithms adopted in the current SVC.

• Journal of Korea Multimedia Society
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• v.19 no.3
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• pp.595-602
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• 2016

With the evolution and development of many kinds of healthcare devices and techniques, u-health standards have emerged as a major issue. Yet, most legacy medical devices and systems are still being used without deployment of the standards. Therefore, it is required to support backward compatibility for u-health standard-compliant systems to communicate with legacy non-standard medical and healthcare devices. This paper proposes a new scheme to support backward compatibility of IEEE 11073 system by adding a codec module to IEEE 11073 agent. The codec converts data sent by non-standard health devices to IEEE 11073 MDER data. Plus, we implemented the proposed IEEE 11073 agent with an Intel Edison board which is one of popular open source H/W platforms. The IEEE 11073 manager of the proposed system can monitor and control legacy non-standard devices through the proposed agent system. In our experimental results, we examined the proposed system can support interoperability between u-health standard and non-standard devices and contribute to the growth and expansion of u-health services.

• The Journal of the Acoustical Society of Korea
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• v.24 no.6
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• pp.358-364
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• 2005

In this Paper. we present a bitrate scalable speech codec which uses an ITU-T G.723.1 as the baseline coder and encodes the synthesis error signal in an enhancement coder. ITU-T P.862 (PESQ) is used to evaluate the Performance of the bitrate scalable coder. Experiments show that 6.7kbps enhancement layer based on G.723.1 5.3kbps produces the increase of 0.39 in MOS and 5.7kbps enhancement layer based on G.723.1 6.3kbps Produces the increase of 0.267 in MOS.

• The Journal of the Acoustical Society of Korea
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• v.16 no.3E
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• pp.56-61
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• 1997

In this paper, we describe a real-time implementation of a multi-channel auido codec system that is based on the MPEG-1 audio algorithm. The major feature of this system is that it has a flexible multi-DSP system that can be adapted for various applications with using up to four TMS320C40 DSPs. The purpose of this paper is to present the problems of the system and is to describe the optimized methods to solve the problems in the view of hardware and software. Our audio codec is composed of an encoder an a decoder system and the bit rate of bitstream is up to 384 kbps. Fast input/output interfaces, DSP overloads, and inter-DSP communications methods with high speed are considered in multi-DSP H/W. Also, to run real-time in S/W, optimizing methods of algorithm are considered. After implementation of system, the subjective assessment method, and 'triple stimulus/hidden reference/double blind' that recommended by ITU-R TG10/3 is adopted for the quality of our system. All test items except one are awarded difference grades(diffgrade) better than 1-. Form the results, multi-channel audio system can be used for HDTV service.

• Journal of Korea Multimedia Society
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• v.18 no.7
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• pp.853-863
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• 2015

In this paper, we proposed cross-layered approach of video codec and communication system for the efficient video streaming service. Conventional video streaming is served by divided system which consist of video codec layer and communication layer. Its disintegration causes the limitation of the performance of video streaming service. With the cross-layered design, each layer could share the information and the service is able to enhance the performance. And we proposed the selective retransmission method in communication system based on the cross-layered system that reflect the information of encoded video data. Selective retransmission method which consider the characteristics of video data improves the performance of video streaming services. We verified the proposed method with raw format full HD test sequence with H.264/AVC codec and MATLAB simulation. The simulation results show that the proposed method improves about 10% PSNR performance.