• Journal of the Institute of Electronics Engineers of Korea SP
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• v.45 no.5
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• pp.35-42
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• 2008

In this paper, we propose a fast coeff_token decoding method based on the re-constructed VLCT. Since the conventional decoding method is still based on large memory accesses, it is not suitable for the multimedia services such as PMP, PMB, DVH-H where fast decoding and low power consumption are required. Based on the analysis for the codeword structure, new structure of the codeword and the corresponding memory architecture are developed in this paper. The simulation results show that the proposed algorithm achieves memory access saving from 10% to 57%, compared to the conventional decoding method. This meant that the issues of tow power consumption and high speed decoding can be resolved without video-quality and coding efficiency degradation.

• Journal of Digital Convergence
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• v.12 no.3
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• pp.269-275
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• 2014

The DAB system was designed to provide CD quality audio and data services for fixed, portable and mobile applications with the required BER below $10^{-4}$. However for the T-DMB system with the video service of MPEG-4 stream, BER should go down $10^{-8}$ by adding FEC blocks which consist of the Reed-Solomon (RS) encoder/decoder and convolutional interleaver/deinterleaver. In this paper we propose two types of turbo coded T-DMB system without altering the puncturing procedure and puncturing vectors defined in the standard T-DMB system for compatibility. One(Type 1) can replace the existing RS code, convolutional interleaver and RCPC code by a turbo code and the other one (Type 2) can substitute the existing RCPC code by a turbo code. Simulation results show that two new turbo coded systems are able to yield considerable performance gain after just 2 iterations. Type 2 system is better than type 1 but the amount of performance improvement is small.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.2A
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• pp.90-97
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• 2005

Generally, DAB systems are divided into two categories, a stand-alone type and a PC/PDA-based type. The PC/PDA-based type has less mobility comparing to the stand-alone type, nevertheless, it has the advantage of using memory, audio/video decoder, or other resources of PC/PDA. The DAB receiver implemented in this paper is a PC-based receiver system employing USB interface. The USB interface bridge is designed using FPGA and EZ-USB development kit and the implemented DAB receiver adopts the bridge and makes use of the stand-alone typed DRK-026 receiver for experiments. The USB interface bridge transforms serial data into USB packets and all of related signals are controlled by hardware logics. The operation of the implemented DAB receiver is verified by sending audio data into the PC for decoding through USB interface bridge.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.1C
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• pp.37-47
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• 2009

When H.264 encoded video streams are transmitted over wireless network, packet loss is unavoidable. Responding on this environment, we propose methods to recover missed motion vector in the decoder: At first, A candidate vector set for missing macroblock is estimated from high correlation coefficient of neighboring motion vectors and missing block vectors the algorithm clusters candidate vectors through distances amongst motion vectors of neighboring blocks. Then the optimal candidate vector is determined by the median value of the clustered motion vector set. In next stage, from the candidate vector set, the final candidate vector of missing block is determined it has minimum distortion value considering directions of neighboring pixels' boundary. Test results showed that the proposed algorithm decreases the candidate motion vectors $23{\sim}61%$ and reduces $3{\sim}4sec$ on average processing(decoding) time comparing the existing H.264 codec. The PSNR, in terms of visual quality is similar to existing methods.

• Journal of Advanced Navigation Technology
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• v.10 no.3
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• pp.262-267
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• 2006

For growing need for the multimedia application in the vehicles, the MOST protocol has been focused on. The MOST protocol supports three kinds of communication modes; short control message, asynchronous packets, and reserved synchronous stream data. Because of a variety of transportation, the MOST is suitable for various applications in vehicle environment. In this paper, we implemented embedded system which is MOST-enabled AMS platform and tested the network communication operation through the control port and the synchronous channel of the source port. We implemented the prototype platforms which communicate each other on the MOST's POF network. Moreover we implemented the DivX decoder attached AMS platform and verified the operation by transferring the video stream and the control messages through the MOST network.

• Journal of the Semiconductor & Display Technology
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• v.17 no.4
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• pp.21-26
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• 2018

Recently, the emergence of 3D cameras, 3D scanners and various cameras including Lidar is expected to be applied to applications such as AR, VR, and autonomous mobile vehicles that deal with 3D data. In Particular, the 3D point cloud data consisting of tens to hundreds of thousands of 3D points is rapidly increased in capacity compared with 2D data, Efficient encoding / decoding technology for smooth service within a limited bandwidth, and efficient service provision technology for differentiating the area of interest and the surrounding area are needed. In this paper, we propose a new quality parameter considering characteristics of 3D point cloud instead of quality change based on assumed video codec in MPEG V-PCC used in 3D point cloud compression, 3D Grid division method and representation for selectively transmitting 3D point clouds according to user's area of interest, and propose a new 3D Manifesto. By using the proposed technique, it is possible to generate more bitrate images, and it is confirmed that the efficiency of network, decoder, and renderer can be increased while selectively transmitting as needed.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.46 no.3
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• pp.68-74
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• 2009

In this paper, we propose an effective memory reduction algorithm to reduce the amount of reference frame buffer and memory bandwidth in video encoder and decoder. In general video codecs, decoded previous frames should be stored and referred to reduce temporal redundancy. Recently, reference frames are recompressed for memory efficiency and bandwidth reduction between a main processor and external memory. However, these algorithms could hurt coding efficiency. Several algorithms have been proposed to reduce the amount of reference memory with minimum quality degradation. They still suffer from quality degradation with fixed-bit allocation. In this paper, we propose an adaptive block-based min-max quantization that considers local characteristics of image. In the proposed algorithm, basic process unit is $8{\times}8$ for memory alignment and apply an adaptive quantization to each $4{\times}4$ block for minimizing quality degradation. We found that the proposed algorithm can obtain around 1.7% BD-bitrate gain and 0.03dB BD-PSNR gain, compared with the conventional fixed-bit min-max algorithm with 37.5% memory saving.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.8C
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• pp.1113-1124
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• 2004

In this paper, we proposed a hardware(H/W) structure which can compress and recontruct the input image in real time operation and implemented it into a FPGA platform using VHDL(VHSIC Hardware Description Language). All the image processing element to process both compression and reconstruction in a FPGA were considered each of them was mapped into H/W with the efficient structure for FPGA. We used the DWT(discrete wavelet transform) which transforms the data from spatial domain to the frequency domain, because use considered the motion JPEG2000 as the application. The implemented H/W is separated to both the data path part and the control part. The data path part consisted of the image processing blocks and the data processing blocks. The image processing blocks consisted of the DWT Kernel fur the filtering by DWT, Quantizer/Huffman Encoder, Inverse Adder/Buffer for adding the low frequency coefficient to the high frequency one in the inverse DWT operation, and Huffman Decoder. Also there existed the interface blocks for communicating with the external application environments and the timing blocks for buffering between the internal blocks The global operations of the designed H/W are the image compression and the reconstruction, and it is operated by the unit of a field synchronized with the A/D converter. The implemented H/W used the 69%(16980) LAB(Logic Array Block) and 9%(28352) ESB(Embedded System Block) in the APEX20KC EP20K600CB652-7 FPGA chip of ALTERA, and stably operated in the 70MHz clock frequency. So we verified the real time operation of 60 fields/sec(30 frames/sec).

• The Journal of the Acoustical Society of Korea
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• v.21 no.6
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• pp.566-575
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• 2002

Low-titrate audio coding enables a number of Internet and mobile multimedia streaming service more efficiently. For the help of next-generation mobile telephone technologies and digital audio/video compression algorithm, we can enjoy the real-time multimedia contents on our mobile devices (cellular phone, PDA notebook, etc). But the limited available bandwidth of mobile communication network prohibits transmitting high-qualify AV contents. In addition, most bandwidth is assigned to transmit video contents. In this paper, we design a novel and simple method for reproducing high frequency components. The spectrum of high frequency components, which are lost by down-sampling, are modeled by the energy rate with low frequency band in Bark scale, and these values are multiplexed with conventional coded bitstream. At the decoder side, the high frequency components are reconstructed by duplicating with low frequency band spectrum at a rate of decoded energy rates. As a result of segmental SNR and MOS test, we convinced that our proposed method enhances the subjective sound quality only 10%∼20% additional bits. In addition, this proposed method can apply all kinds of frequency domain audio compression algorithms, such as MPEG-1/2, AAC, AC-3, and etc.

• Journal of Broadcast Engineering
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• v.11 no.3 s.32
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• pp.363-378
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• 2006

The Digital Multimedia Broadcasting(DMB) system is developed to offer high quality audio-visual multimedia contents to the mobile environment. The system adopts MPEG-4 standard for the main video, audio and other media format. It also adopts the MPEG-4 scene description for interactive multimedia contents. The animated and interactive contents can be actualized by BIFS(Binary Format for Scene), the binary format for scene description that refers to the spatio-temporal specifications and behaviors of the individual objects. As more interactive contents are, the scene description is also needed more high bitrate. However, the bandwidth for allocating meta data such as scene description is restrictive in mobile environment. On one hand, the DMB terminal starts demultiplexing content and decodes individual media by its own decoder. After decoding each media, rendering module presents each media stream according to the scene description. Thus the BIFS stream corresponding to the scene description should be decoded and parsed in advance of presenting media data. With these reason, the transmission delay of BIFS stream causes the delay of whole audio-visual scene presentation although the audio or video streams are encoded in very low bitrate. This paper presents the effective optimization technique for adapting BIFS stream into expected MPEG-2 TS bitrate without any bandwidth waste and avoiding the transmission delay of the initial scene description for interactive DMB contents.