• The Journal of the Acoustical Society of Korea
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• v.19 no.4
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• pp.37-44
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• 2000

In this paper, we described the real-time implementation effort of MPEG-4 audio HVXC (Harmonic Vector eXcitation Coding) algorithm for very low bitrates, which has target applications from mobile communications to Internet telephony, on current high performance floating point TMS320C6701 DSP. We adopted a hardware structure for real-time operation. In order for software optimization, we used C- and assembly-language level optimizations for time-critical functional codes. Utilizing the internal program memory of the DSP as the program cache, the internal data memory overlap technique and DMA functionality, we could get a goal of realtime operation of HVXC codec both at 2 kbit/s and at 4 kbit/s. For an encoder at 2 kbit/s, the optimization ratio to original code is about 96 %. Finally, we got the subjective quality of MOS 2.45 at 2 kbit/s from an informal quality test.

• Journal of Broadcast Engineering
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• v.25 no.5
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• pp.709-722
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• 2020

Point Cloud which is a cluster of numerous points can express 3D object beyond the 2D plane. Each point contains 3D coordinate and color data basically, reflectance or etc. additionally. Point Cloud demand research and development much higher effective compression technology. Video-based Point Cloud Compression (V-PCC) technology in development and standardization based on the established video codec. Despite its high effective compression technology, point cloud service will be limited by terminal spec and network conditions. 2D video had the same problems. To remedy this kind of problem, 2D video is using Scalable High efficiency Video Coding (SHVC), Dynamic Adaptive Streaming over HTTP (DASH) or diverse technology. This paper proposed a density scalability method using SHVC codec in V-PCC.

• The Journal of the Acoustical Society of Korea
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• v.15 no.2
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• pp.55-60
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• 1996

This paper describes the subjective assessment of the multi-channel audio quality according to the bitrate changing and evaluates the predistortion effect to avoid the unmasked noise after matrixing/dematrxing process in transmission and regeneration of the multi-channel audio. The simulation is processed by the perceptual coding that is MPEG-2 Audio layer II algorithm. We evaluate the quality improvement about predistortion using or not by 384, 320, 256, 128kbps. As the result of the double blind subjective assessment, 5 Grade-Impairment Scale is scored under minus one to 320kbps and so audio quality is evaluated to be perceptible, but not annoying in 3/2 channel. The effect of the predistortion is improved one level in 128kbps and especially speech test material I better improved than music test materials.

• The KIPS Transactions:PartA
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• v.18A no.2
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• pp.39-44
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• 2011

This paper proposes the new high-performance circuit architecture of the transform and quantization for unified video CODEC. The proposed architecture can be applied to all kinds of transforms and quantizations for the video compression standards such as JPEG, MPEG-1/2/4, H.264 and VC-1. We defined the permutation matrices to reorder the transform matrix of the $8{\times}8$ DCT and partitioned the reordered $8{\times}8$ transform matrix into four $4{\times}4$ sub-matrices. The $8{\times}8$ DCT is performed by repeating the $4{\times}4$ DCT's based on the reordered and partitioned transform matrices. Since our circuit accepts the transform coefficients from the users, it can be extended very easily to cover any kind of DCT-based transforms for future standards. The multipliers in the DCT circuit are shared by the quantization circuit in order to minimize the circuit size. The quantization circuit is merged into the DCT circuit without any significant increase of circuit resources and processing time. We described the proposed DCT and quantization circuit at RTL, and verified its operation on FPGA board.

• Journal of Broadcast Engineering
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• v.22 no.6
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• pp.829-849
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• 2017

Due to the large-scale TV display, the convergence of broadcasting and broadband, and the advancement of signal compression and transmission technology, terrestrial digital broadcasting has evolved into UHD broadcasting capable of providing simultaneous broadcasting of fixed UHD and mobile HD. The Korean standard for terrestrial UHDTV broadcasting is based on ATSC 3.0, the broadcasting standard of North America. The terrestrial UHDTV broadcasting standard chose that as a new AV codec standard, HEVC video codec which can compress with higher efficiency compared to AVC, and MPEG-H 3D audio codec for realistic audio. Also, DASH and MMT are adopted as transmission format instead of MPEG-2 TS to support broadband as well as broadcasting network, and in order to provide 4K UHD/mobile HD service simultaneously ROUTE multiplexing technology is applied. In this paper, we propose an audio/video encoder, which is required to provide HDR/WCG supported high quality video service, 10.2 channel/4 object supporting stereo sound service, fixed UHD and mobile HD simultaneous broadcasting service based on ATSC3.0, also we implemented the ATSC 3.0 LDM system for ROUTE/DASH packager, multiplexing system and physical layer transmission/reception, and verified the service ability by applying it to real time broadcast environment.

• Journal of KIISE:Computer Systems and Theory
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• v.32 no.2
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• pp.95-105
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• 2005

This paper proposes a new dynamic video adaptation scheme that could generate an adapted video stream customized to the requesting mobile device and current network status without repeated decode-encode cycles. In the proposed adaptation scheme, the characteristics of the video codec such as MPEG-1/-2/-4 are analyzed in advance focused on the relationships between the size and Quality of the encoded video stream, and they are stored in the proxy as a codec-dependent characteristic table. When a mobile device requests a video stream, it is dynamically decoded-encoded in the proxy with the highest quality to extract the contents-dependent attributes of the requested video stream. By comparing these attributes with codec-dependent characteristic table, the size and Quality of the requested video stream when being adapted to the target mobile device could be predicted. With this prediction, a version of adapted video stream, that meets the size constraints of mobile device while keeping the quality of encoded video stream as high as possible, could be selected without repeated decode-encode cycles. Experimental results show that the errors in our proposed scheme are less than 5% and produce an appropriate adapted video stream very quickly. It could be used t(1 build a proxy server for mobile devices that could quickly transcode the video streams widely spread in Internet which are encoded with various video codecs.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.19 no.2
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• pp.386-392
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• 2015

Data in a image compression codec are processed with a specific regular block size. The processing order of block sized data is changed in specific function blocks and the data is packed in memory and read by a new sequence. To maintain a regular throughput rate, double buffering is normally used that interleaving two block sized memory to do concurrent read and write operations. Single buffering using only one block sized memory can be adopted to the simple data reordering, but when a complicate reordering occurs, irregular address changes prohibit from implementing adequate address generating for single buffering. This paper shows that there is a predictable and recurring regularity of changing address access orders within a finite updating counts and suggests an effective method to implement. The data reordering function using suggested idea is designed with HDL and implemented with TSMC 0.18 CMOS process library. In various scan blocks, it shows more than 40% size reduction compared with a conventional method.

• The Transactions of the Korea Information Processing Society
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• v.6 no.2
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• pp.534-542
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• 1999

A sell-designed rate control strategy can improve overall picture quality for video transmission over a constant bit rate channel and the rate control method is not a normative part of MPEG-video standard, the performance of MPEG video codec can be quite different depends on how to implement the rate control scheme. The rate control scheme proposed in MPEG show good results when scene changes is not occurred. But it has weakness that it does not properly handle scene-changed pictures. Therefore picture quality after scene change is deteriorated, and possibility of overflow occurrence becomes high. In this paper, a new method for detection of scene change occurrence using local variance and a new determination scheme for adaptive quantization parameter, mqunt, which can consider local characteristic of an image by using previously computed the local variance from the scene change detection part are proposed. IN addition, and adaptive rate control scheme which can handles scene changed picture very efficiently by scene-changed picture is proposed. Computer simulations are performed to verify the performance of the proposed algorithm. The suggested detection algorithm precisely detected scene change. And the proposed rate control scheme shows better rate control performance as compared with that of the conventional MPEG scheme.

• Proceedings of the KOSOMBE Conference
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• v.1995 no.05
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• pp.235-238
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• 1995

In this paper, we propose an effective coding method of the angiography by using the scalable structure in the frequency domain for MPACS(Medical Picture Archiving and Communication System). We employed the subband decomposition method and MPEG-2 system which is the international standard coding method of the general moving picture. After the subband decomposition is applied to split an input image into 4 bands in the spatial frequency domain, the motion compensated DPCM coding method of MPEG-2 is carried out for each subband. As a result, an easily controllable coding Structure is accomplished by composing the compound hit stream for each subband group. Follows are the simulation results of the proposed sheme for the angiography. A scalable structure which can be easily controlled for a loss of transmission or the band limit can be accomplisbed in the MPEG-2 stucture by the subband decomposition minimizing the side information. And by reducing the search area of the motion vector between -4 and 3, the processing speed of a codec is enhanced by more than two times without a loss of the picture quality compare with the conventional DCT coefficients decompositon method. And the processing speed is considerably improved in the case of the parallel construction of each subband in the hardware.

• Journal of the Institute of Electronics Engineers of Korea CI
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• v.48 no.1
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• pp.116-126
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• 2011

In this paper, we present a multi-media processor which can decode multiple-format video standards. The designed processor is evaluated with optimized MPEG-2, MPEG-4, and AVS (Audio video standard). There are two approaches for developing of real-time video decoders. First, hardware-based system is much superior to a processor-based one in execution time. However, it takes long time to implement and modify hardware systems. On the contrary, the software-based video codecs can be easily implemented and flexible, however, their performance is not so good for real-time applications. In this paper, in order to exploit benefits related to two approaches, we designed a processor called ASIP(Application specific instruction-set processor) for video decoding. In our work, we extracted eight common modules from various video decoders, and added several multimedia instructions to the processor. The developed processor for video decoders is evaluated with the Synopsys platform simulator and a FPGA board. In our experiment, we can achieve about 37% time saving in total decoding time.