• Journal of Broadcast Engineering
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• v.27 no.4
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• pp.548-560
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• 2022

Video based Point Cloud Compression (V-PCC) is one of the compression methods for compressing point clouds, and shows high efficiency in dynamic point cloud compression with movement due to the feature of compressing point cloud data using an existing video codec. Accordingly, V-PCC is drawing attention as a core technology for immersive content services such as AR/VR. In order to effectively service these V-PCC contents through a media streaming platform, it is necessary to encapsulate them in the existing media file format, ISO based Media File Format (ISOBMFF). However, in order to service through an adaptive streaming platform such as Dynamic Adaptive Streaming over HTTP (DASH), it is necessary to encode V-PCC contents of various qualities and store them in the server. Due to the size of the 2D media, it causes a great burden on the encoder and the server compared to the existing 2D media. As a method to solve such a problem, it may be considered to configure a streaming platform based on content obtained through V-PCC content encoding based on SHVC. Therefore, this paper encapsulates the SHVC-based V-PCC bitstream into ISOBMFF suitable for DASH service and proposes a configuration method to service it. In addition, in this paper, we propose ISOBMFF encapsulation and DASH configuration method to effectively service SHVC-based V-PCC contents, and confirm them through verification experiments.

• Journal of the Institute of Electronics Engineers of Korea SD
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• v.39 no.3
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• pp.98-109
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• 2002

Bluetooth technology is a publicly available specification proposed for Radio Frequency (RF) communication for short-range :1nd point-to-multipoint voice and data transfer. It operates in the 2.4㎓ ISM(Industrial, Scientific and Medical) band and offers the potential for low-cost, broadband wireless access for various mobile and portable devices at range of about 10 meters. In this paper, we describe the structure and the test results of the bluetooth baseband module with direct memory access method we have developed. This module consists of three blocks; link controller, UART interface, and audio CODEC. This module has a bus interface for data communication between this module and main processor and a RF interface for the transmission of bit-stream between this module and RF module. The bus interface includes DMA interface. Compared with the link controller with FIFOs, The module with DMA has a wide difference in size of module and speed of data processing. The small size module supplies lorr cost and various applications. In addition, this supports a firmware upgrade capability through UART. An FPGA and an ASIC implementation of this module, designed as soft If, are tested for file and bit-stream transfers between PCs.

• Journal of the Institute of Electronics Engineers of Korea SD
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• v.44 no.11
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• pp.101-108
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• 2007

In this paper, we propose and verify an adaptive block watermarking algorithm based on JPEG2000 DWT, which determines watermarking for the original image by two scaling factors in order to overcome image degradation and blocking problem at the edge. Adaptive block watermarking algorithm uses 2 scaling factors, one is calculated by the ratio of present block average to the next block average, and the other is calculated by the ratio of total LL subband average to each block average. Signals of adaptive block watermark are obtained from an original image by itself and the strength of watermark is automatically controlled by image characters. Instead of conventional methods using identical intensity of a watermark, the proposed method uses adaptive watermark with different intensity controlled by each block. Thus, an adaptive block watermark improves the visuality of images by 4$\sim$14dB and it is robust against attacks such as filtering, JPEG2000 compression, resizing and cropping. Also we implemented the algorithm in ASIC using Hynix 0.25${\mu}m$ CMOS technology to integrate it in JPEG2000 codec chip.

• Journal of Advanced Navigation Technology
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• v.17 no.1
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• pp.69-79
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• 2013

In this paper, DCT(Discrete Cosine Transform) and CAVLC(Context Adaptive Variable Length Coding) are co-designed as hardware IP with software operation of the other modules in H.264/AVC codec. In order to increase the operation speed, a new method using SHIFT table is proposed. As a result, enhancement of about 16(%) in the operation speed is obtained. Designed Hardware IPs are downloaded into Virtex-4 FX60 FPGA in the ML-410 development board and H.264/AVC encoding is performed with Microblaze CPU implemented in FPGA. Software modules are developed from JM13.2 to make C code. In order to verify the designed Hardware IPs, Modelsim program is used for functional simulation. As a result that all Hardware IPs and software modules are downloaded into the FPGA, improvement of processing speed about multiples of 16 in case of DCT hardware IP and multiples of 10 in case of CAVLC compared with software-only processing. Although this paper deals with co-design of H/W and S/W for H.264, it can be utilized for the other embedded system design.

• Journal of Broadcast Engineering
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• v.15 no.5
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• pp.653-664
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• 2010

Generally existing video codec employs entropy coding to deal with residual signals with considering temporal and spatial properties. Scalable Video Coding(SVC) which is extension of H.264/AVC has three technical concepts for removing redundancies between inter-layers. In spite of using novel prediction method between inter-layers in SVC, it is still using same entropy coding method to residual signals. According to the studies, the residual obtained by inter-layer prediction technique has different features of residual signal acquired by spatial or temporal prediction technique. In this paper, we propose an efficient entropy coding method which codes the residual signal obtained by inter-layer prediction with regarding its features adequately. We re-designed the Coded Block Pattern(CBP) table suitably for inter-layer texture prediction. The experiments show that the proposed method can further reduce the BD-Bitrate up to average 2.20% in 4CIF and 1.14% in CIF resolution compared to the existing JSVM 9.18.

• Journal of Broadcast Engineering
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• v.13 no.5
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• pp.732-743
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• 2008

In H.264/AVC, 4$\times$4 block transform is used for intra and inter prediction instead of 8$\times$8 block transform. Using small block size coding, H.264/AVC obtains high temporal prediction efficiency, however, it has limitation in utilizing spatial redundancy. Motivated on these points, we propose a multi-dimensional transform which achieves both the accuracy of temporal prediction as well as effective use of spatial redundancy. From preliminary experiments, the proposed multi-dimensional transform achieves higher energy compaction than 2-D DCT used in H.264. We designed an integer-based transform and quantization coder for multi-dimensional coder. Moreover, several additional methods for multi-dimensional coder are proposed, which are cube forming, scan order, mode decision and updating parameters. The Context-based Adaptive Variable-Length Coding (CAVLC) used in H.264 was employed for the entropy coder. Simulation results show that the performance of the multi-dimensional codec appears similar to that of H.264 in lower bit rates although the rate-distortion curves of the multi-dimensional DCT measured by entropy and the number of non-zero coefficients show remarkably higher performance than those of H.264/AVC. This implies that more efficient entropy coder optimized to the statistics of multi-dimensional DCT coefficients and rate-distortion operation are needed to take full advantage of the multi-dimensional DCT. There remains many issues and future works about multi-dimensional coder to improve coding efficiency over H.264/AVC.

• Journal of Broadcast Engineering
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• v.24 no.2
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• pp.306-314
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• 2019

Since the resolutions of videos increase rapidly, there are continuing needs for effective video compression methods despite an increase in the transmission bandwidth. In order to satisfy such a demand, a reconstructive video coding (RVC) method by using a super resolution has been proposed. Since RVC reduces the resolution of the input video, when frames are compressed to the same size, the number of bits per pixel increases, thereby reducing coding artifacts caused by video coding. However, RVC method using super resolution is not effective in all target bitrates. Comparing the size of the loss generated while downsizing the resolution and the size of the loss caused by the video compression, only when the size of loss generated in the video compression is larger, RVC method can perform the improved compression performance compared to direct video coding. In particular, since HEVC has considerably higher compression performance than the previous standard video codec, it can be experimentally confirmed that the compression distortions become larger than the distortions of downsizing the resolution only in the very low-bitrate conditions. In this paper, we applied RVC based HEVC in various video types and measured the target bitrates that RVC method can be effectively applied.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.40 no.5
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• pp.397-409
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• 2003

This paper describes an implementation of radix-4 trellis parallel architecture and backward state transition control trace back Viterbi decoder, and presents the application results to high speed wireless LAN. The radix-4 parallelized architecture Vietrbi decoder can not only improve the throughput with simple structure, but also have small processing delay time and overhead circuit compared to M-step trellis architecture one. Based on these features, this paper addresses a novel Viterbi decoder which is composed of branch metric computation, architecture of ACS and trace back decoding by sequential control of backward state transition for the implementation of radix-4 trellis parallelized structure. With the proposed architecture, the decoding of variable code rate due to puncturing the base code can easily be implemented by the unified Viterbi decoder. Moreover, any additional circuit and/or peripheral control logic are not required in the proposed decoder architecture. The trace back decoding scheme with backward state transition control can carry out the sequential decoding according to ACS cycle clock without additional circuit for survivor memory control. In order to evaluate the usefulness, the proposed method is applied to channel CODEC of the IEEE 802.11a high speed wireless LAN, and HDL coding simulation results are presented.

• 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.

• Journal of Intelligence and Information Systems
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• v.24 no.2
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• pp.21-35
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• 2018

Since the introduction of MP3 players, CD recordings have gradually been vanishing, and the music consuming environment of music users is shifting to mobile devices. The introduction of smart devices has increased the utilization of music through music playback, mass storage, and search functions that are integrated into smartphones and tablets. At the time of initial MP3 player supply, the bitrate of the compressed music contents generally was 128 Kbps. However, as increasing of the demand for high quality music, sound quality of 384 Kbps appeared. Recently, music content of FLAC (Free License Audio Codec) format using lossless compression method is becoming popular. The download service of many music sites in Korea has classified by unlimited download with technical protection and limited download without technical protection. Digital Rights Management (DRM) technology is used as a technical protection measure for unlimited download, but it can only be used with authenticated devices that have DRM installed. Even if music purchased by the user, it cannot be used by other devices. On the contrary, in the case of music that is limited in quantity but not technically protected, there is no way to enforce anyone who distributes it, and in the case of high quality music such as FLAC, the loss is greater. In this paper, the author proposes an audio watermarking technology for copyright protection of high quality stereo music. Two kinds of information, "Copyright" and "Copy_free", are generated by using the turbo code. The two watermarks are composed of 9 bytes (72 bits). If turbo code is applied for error correction, the amount of information to be inserted as 222 bits increases. The 222-bit watermark was expanded to 1024 bits to be robust against additional errors and finally used as a watermark to insert into stereo music. Turbo code is a way to recover raw data if the damaged amount is less than 15% even if part of the code is damaged due to attack of watermarked content. It can be extended to 1024 bits or it can find 222 bits from some damaged contents by increasing the probability, the watermark itself has made it more resistant to attack. The proposed algorithm uses quantization in DCT so that watermark can be detected efficiently and SNR can be improved when stereo music is converted into mono. As a result, on average SNR exceeded 40dB, resulting in sound quality improvements of over 10dB over traditional quantization methods. This is a very significant result because it means relatively 10 times improvement in sound quality. In addition, the sample length required for extracting the watermark can be extracted sufficiently if the length is shorter than 1 second, and the watermark can be completely extracted from music samples of less than one second in all of the MP3 compression having a bit rate of 128 Kbps. The conventional quantization method can extract the watermark with a length of only 1/10 compared to the case where the sampling of the 10-second length largely fails to extract the watermark. In this study, since the length of the watermark embedded into music is 72 bits, it provides sufficient capacity to embed necessary information for music. It is enough bits to identify the music distributed all over the world. 272 can identify $4*10^{21}$, so it can be used as an identifier and it can be used for copyright protection of high quality music service. The proposed algorithm can be used not only for high quality audio but also for development of watermarking algorithm in multimedia such as UHD (Ultra High Definition) TV and high-resolution image. In addition, with the development of digital devices, users are demanding high quality music in the music industry, and artificial intelligence assistant is coming along with high quality music and streaming service. The results of this study can be used to protect the rights of copyright holders in these industries.