• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.18 no.12
• /
• pp.1956-1969
• /
• 1993

The tilted Huffman trees are used in JPEG and MPEG image compression standards for Huffman coding. In this paper we propose a new decoding technique for Huffman code, symbols are decoded by repeatedly obtaining the predefined number of consecutive bits and accessing symbol tables based on the obtaining bits. We show that the size of the symbol table can be small if the Huffman tree is tilted. Specifically, we show an upper bound on the size in this paper. Since the proposed method processes multiple bits at each clock, it can be used for real time processing. We show such evaluation results.

• Journal of Computing Science and Engineering
• /
• v.4 no.3
• /
• pp.207-224
• /
• 2010

This paper presents a compression scheme for mesh geometry, which is suitable for mobile graphics. The main focus is to enable real-time decoding of compressed vertex positions while providing reasonable compression ratios. Our scheme is based on local quantization of vertex positions with mesh partitioning. To prevent visual seams along the partitioning boundaries, we constrain the locally quantized cells of all mesh partitions to have the same size and aligned local axes. We propose a mesh partitioning algorithm to minimize the size of locally quantized cells, which relates to the distortion of a restored mesh. Vertex coordinates are stored in main memory and transmitted to graphics hardware for rendering in the quantized form, saving memory space and system bus bandwidth. Decoding operation is combined with model geometry transformation, and the only overhead to restore vertex positions is one matrix multiplication for each mesh partition. In our experiments, a 32-bit floating point vertex coordinate is quantized into an 8-bit integer, which is the smallest data size supported in a mobile graphics library. With this setting, the distortions of the restored meshes are comparable to 11-bit global quantization of vertex coordinates. We also apply the proposed approach to compression of vertex attributes, such as vertex normals and texture coordinates, and show that gains similar to vertex geometry can be obtained through local quantization with mesh partitioning.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.27 no.7A
• /
• pp.659-667
• /
• 2002

In this dissertation, in the Turbo Code used for error correction coding of the recent digital communication systems, we propose a new S-R interleaver that has the better performance than the existing block interleaver, and the Turbo Decoder that has the parallel concatenated New structure using the MAP algorithm. For real-time voice and video services over the third generation mobile communications, the performance of two proposed methods is analyzed by the reduced decoding delay using the variable decoding method by computer simulation over multipath channels of DS-CDMA system. Also, a Modified NCMA based on conventional NCMA is proposed to improve the channel efficiency in the mobile communication system, and is investigated over the multi-user environment of DS-CDMA system through computer simulation.

• The Transactions of the Korea Information Processing Society
• /
• v.4 no.11
• /
• pp.2671-2678
• /
• 1997

The MHEG-5 standard defines representation of multimedia and hypermedia information. The MHEG-5 can make it possible to interchange and present information between heterogeneous networked environment. Also, it can provide real-time user interaction. MHEG-5 object generating is the essential requirements for the MHEG-5 applications, In this paper, we designed encoding/decoding class library for the MHEG-5 objects. We can use it for the interchange on the network. The designed class library can provide base technology for the MHEG-5 based distributed multimedia and hypermedia services on the networked environment.

• The Transactions of the Korea Information Processing Society
• /
• v.4 no.3
• /
• pp.880-894
• /
• 1997

In this paper,we propose a new algorithm to perform DCT(Discrete Cosine Transform) withn the area reduced by prdeicting position of quantization coefficients to be zero.This proposed algorithm not only decreases the enoding time and the decoding time by reducing computation amount of FDCT(Forward DCT)and IDCT(Inverse DCT) but also increases comprossion ratio by performing each diffirent horizontal- vereical zig-zag scan assording to the calssified block size for each block on the huffiman coeing.Traditional image coding method performs the samd DCT computation and zig-zag scan over all blocks,however this proposed algorthm reduces FDCT computation time by setting to zero insted of computing DCT for quantization codfficients outside classfified block size on the encoding.Also,the algorithm reduces IDCT computation the by performing IDCT for only dequantization coefficients within calssified block size on the decoding.In addition, the algorithm reduces Run-Length by carrying out horizontal-vertical zig-zag scan approriate to the slassified block chraateristics,thus providing the improverment of the compression ratio,On the on ther hand,this proposed algorithm can be applied to 16*16 block processing in which the compression ratio and the image resolution are optimal but the encoding time and the decoding time take long.Also,the algorithm can be extended to motion image coding requirng real time processing.

• The Journal of the Acoustical Society of Korea
• /
• v.32 no.4
• /
• pp.346-353
• /
• 2013

This paper deals with the implementation of a high-quality audio system for karaoke. For improving the key/tempo changes performance, we separated the audio into many musical instrument channels. By separating musical instrument channels, high-quality key/tempo changes can be achieved and we confirmed this using the cross-correlation distribution and the MOS evaluation. The improved audio system was implemented using the TMS320C6747 DSP with fixed/floating-point operations. The implemented audio system can perform the multi-channel WMA decoding, the MP3 encoding/decoding, the wav playing, the EQ, and the key/tempo changes in real time. The WMA channels used for processing the separated instrument channels. The audio system includs the MP3 encoding/decoding function for playing and recording and the wav channel for the effect sound.

• Journal of the Korea Institute of Information and Communication Engineering
• /
• v.7 no.8
• /
• pp.1854-1863
• /
• 2003

In the recent digital communication systems, the performance of Turbo Code using the mr correction coding depends on the interleaver influencing the free distance determination and the recursive decoding algorithms that is executed in the huh decoder. However, performance depends on the interleaver depth that needs many delays over the reception process. Moreover, turbo code has been blown as the robust coding methods with the confidence over the fading channel. International Telecommunication Union(ITU) has recently adopted it as the standardization of the channel coding over the third generation mobile communications(IMT­2000). Therefore, in this paper, we preposed the interleaver that has the better performance than existing block interleaver, and modified turbo decoder that has the parallel concatenated structure using MAP algorithm. In the real­time voice and video service over third generation mobile communications, the performance of the proposed two methods was analyzed and compared with the existing methods by computer simulation in terms of reduced decoding delay using the variable decoding method over AWGN and fading channels for CDMA environments.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.37C no.12
• /
• pp.1281-1288
• /
• 2012

This paper presents the design and implementation of a multi-mode mobile broadcasting Emergency Warning System (EWS) transmission and reception system which enables the propagation of disaster information using the encoding and transmission, reception and decoding methods specified in diverse mobile broadcasting standards. The implemented system supports global mobile broadcasting standards such as Terrestrial Digital Multimedia Broadcasting (T-DMB), Digital Video Broadcasting-Handheld (DVB-H), Integrated Services Digital Broadcasting-Terrestrial (ISDB-T), and the Digital Radio Mondiale (DRM) digital radio standard. The system consists of two key part: an encoding/transmission part and a reception/decoding part. The multi-mode mobile broadcasting EWS encoding and transmission system generates EWS data according to each broadcasting specification. The generated EWS data is then transmitted through a channel interface which meets the commercial broadcasting equipment specification. The receiver system receives and decodes the EWS data on a single hardware platform and can display the results on screen. Verification and conformity testing has been carried out on the implemented system by transmitting emergency data for each mode in real-time and displaying the received information in text on the receiver display.

• Proceedings of the Korea Institute of Convergence Signal Processing
• /
• 2000.12a
• /
• pp.13-16
• /
• 2000

In the recent mobile communication systems, the performance of Turbo Code using the error correction coding depends on the interleaver influencing the free distance determination and the recursive decoding algorithms that is executed in the turbo decoder. However, performance depends on the interleaver depth that need great many delay over the reception process. Moreover, Turbo Code has been known as the robust coding methods with the confidence over the fading channel. The International Telecommunication Union(ITU) has recently adopted as the standardization of the channel coding over the third generation mobile communications the same as IMT-2000. Therefore, in this paper, we proposed of that has the better performance than existing Turbo Decoder that has the parallel concatenated four-step structure using MAP algorithm. In the real-time voice and video service over the third generation mobile communications, the performance of the proposed method was analyzed by the reduced decoding delay using the variable decoding method by computer simulation over AWGN and lading channels.

• Journal of Broadcast Engineering
• /
• v.6 no.3
• /
• pp.260-269
• /
• 2001

This paper proposes an efficient algorithm extracting anchor ranges that exist in news video for the unit structuring of news. To this purpose, this paper uses anchors face in the frame rather than the cuts where the scene changes are occurred. In anchor range, we find the end position (frame) of anchor range with the FRFD(Face Region Frame Difference). On the other hand, in not-anchor range, we find the start position of anchor range by extracting anchors face. The process of extracting anchors face is consists of two parts to enhance the computation time for WPEG decoding. The first pact is to find candidates of anchors face through rough analysis with partial decoding MPEG and the second part is to verify candidates of anchors face with fully decoding. It is possible to use the result of this process in basic step of news analysis. Especially, the fast processing and the high recall rate of this process are suitable to apply for the real news service.