• Journal of Communications and Networks
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• v.16 no.3
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• pp.270-279
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• 2014

The Viterbi decoding algorithm, which provides maximum - likelihood decoding, is currently considered the most widely used technique for the decoding of codes having a state description, including the class of linear error-correcting convolutional codes. Two classes of nonbinary convolutional codes are presented. Distance preserving mapping convolutional codes and M-ary convolutional codes are designed, respectively, from the distance-preserving mappings technique and the implementation of the conventional convolutional codes in Galois fields of order higher than two. We also investigated the performance of these codes when combined with a multiple frequency-shift keying (M-FSK) modulation scheme to correct narrowband interference (NBI) in power-line communications channel. Themodification of certain detectors of the M-FSK demodulator to refine the selection and the detection at the decoder is also presented. M-FSK detectors used in our simulations are discussed, and their chosen values are justified. Interesting and promising obtained results have shown a very strong link between the designed codes and the selected detector for M-FSK modulation. An important improvement in gain for certain values of the modified detectors was also observed. The paper also shows that the newly designed codes outperform the conventional convolutional codes in a NBI environment.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2009.01a
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• pp.351-354
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• 2009

Distributed video coding (DVC) is a new coding paradigm that enables to exploit the statistics among sources only in decoder and to achieve extremely low complex video encoding without any loss of coding efficiency. Wyner-Ziv coding, a particular implementation of DVC, reconstructs video by correcting noise on side information using channel code. Since a good quality of side information brings less noise to be removed by the channel code, generation of good side information is very important for the overall coding efficiency. However, if there are complex motions among frames, it is very hard to generate a good quality of side information without any information of original frame. In this paper, we propose a method to enhance the quality of the side information using small amount of additional information of original frame in the form of hash. By decoder's informing encoder where the hash has to be transmitted, side information can be improved enormously with only small amount of hash data. Therefore, the proposed method gains considerable coding efficiency. Results of our experiment have verified average PSNR gain up to 1 dB, when compared to the well-known DVC codec, known as DISCOVER codec.

• 한국정보통신설비학회:학술대회논문집
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• 2005.08a
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• pp.417-421
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• 2005

In the recent digital 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 needs many delays over the reception process. Moreover, turbo code has been known 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(IMT-2000). Therefore, in this paper, we proposed 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 real-time 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.

• Journal of IKEEE
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• v.13 no.4
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• pp.110-115
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• 2009

The motion compensation is the most time-consuming and complex unit in the H.264 decoder. The performance of the motion compensation is determined by the calculation of pixel interpolation. The quarter-pixel interpolation is achieved using 6-tap horizontal or vertical FIR filters for luminance data and bilinear FIR filters for chroma data. We propose the architecture for interpolation of luminance and chroma data in H.264 decoders. It is composed of dual-channel pipelined processing elements and can interpolate integer-, half- and quarter-pixel data. The number of the processing cycles is different depending on the position. The processing elements are composed of adders and shifters to reduce the complexity while the accuracy of the pixel data are maintained. We design interpolators for luminance and chroma data using Verilog-HDL and verify the function and performance by implementing using an FPGA.

• The Journal of the Acoustical Society of Korea
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• v.34 no.4
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• pp.295-302
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• 2015

Acoustic channels are characterized by long multipath spreads that cause inter-symbol interference. The way in which this fact influences the design of the receiver structure is considered. To satisfy performance and throughput, we presented consecutive iterative BCJR (Bahl, Cocke, Jelinek, Raviv) equalization to improve the performance and throughput. To achieve low error performance, we resort to powerful BCJR equalization algorithms that iteratively update probabilistic information between inner decoder and outer decoder. Also, to achieve high throughput, we divide long packet into consecutive small packets, and the estimate channel information of previous packets are compensated to next packets. Based on experimental channel response, we confirmed that the performance is improved for long length packet size.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.7 no.8
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• pp.1854-1863
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• 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 KIPS Transactions:PartD
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• v.14D no.7
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• pp.801-808
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• 2007

Video coding standards such as MPEG-2, MPEG-4, H.263, and H.264 transmit a compressed video data using wired/wireless communication line with limited bandwidth. Because highly compressed bit-streams is likely to fragile to error from channel noise, video is damaged by error. There have been many research works on error concealment techniques, which recover transmission errors at decoder side [1, 2]. We designed an error concealment technique for lost motion vectors of H.264 video coding. In this paper, we propose a Kalman filter based motion vector recovery scheme, and experimented with standard video sequences. The experimental results show that our scheme restores original motion vector with more precision of 0.91 - 1.12 on average over conventional H.264 decoding with no error recovery.

• The Journal of Korean Institute of Information Technology
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• v.17 no.3
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• pp.31-41
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• 2019

In this paper, we design and implement the simulator for the transmitter and receiver of 4D-8PSK TCM with 2.0, 2.25, 2.5, and 2.75 bits/symbol-channel transmission efficiency recommended by the CCSDS for satellite communications, and then analyze the BER performance of 4D-8PSK TCM system in AWGN channel. The transmitter of 4D-8PSK TCM is designed in accordance with the recommendation in the CCSDS standard. Meanwhile, for the receiver design of 4D-8PSK TCM, we design the differential decoder generalizing the differential encoder/decoder scheme. The trellis decoding algorithm is designed by applying the auxiliary trellis information and the Viterbi algorithm, and an 8-dimensional constellation mapper equation given in the CCSDS standard is deconstructed to design constellation mapper. Especially, we present the optimized receiver for 4D-8PSK TCM system by investigating the BER performances for the traceback lengths in the Viterbi decoder through computer simulations..

• The Journal of Korean Institute of Communications and Information Sciences
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• v.42 no.2
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• pp.432-443
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• 2017

The primary goal of the MPEG-H 3D Audio standard is to provide immersive audio environments for high-resolution broadcasting services such as UHDTV. This standard incorporates a wide range of technologies such as encoding/decoding technology for multi-channel/object/scene-based signal, rendering technology for providing 3D audio in various playback environments, and post-processing technology. The reference software decoder of this standard is a structure combining several modules and can operate in various modes. Each module is composed of independent executable files and executed sequentially, real time decoding is impossible. In this paper, we make DLL library of the core decoder, format converter, object renderer, and binaural renderer of the standard and integrate them to enable frame-based decoding. In addition, by measuring the computation complexity of each mode of the MPEG-H 3D-Audio decoder, this paper also provides a reference for selecting the appropriate decoding mode for various hardware platforms. As a result of the computational complexity measurement, the low complexity profiles included in Korean broadcasting standard has a computation complexity of 2.8 times to 12.4 times that of the QMF synthesis operation in case of rendering as a channel signals, and it has a computation complexity of 4.1 times to 15.3 times of the QMF synthesis operation in case of rendering as a binaural signals.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.4C
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• pp.365-373
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• 2003

In this paper, we design a versatle RS decoder which can decode RS codes of any block length n as well as any message length k, based on a modified Euclid's algorithm (MEA). This unique feature is favorable for a shortened RS code of any block length it eliminates the need to insert zeros before decoding a shortened RS code. Furthermore, the value of error correcting capability t can be changed in real time at every codeword block. Thus, when a return channel is available, the error correcting capability can be adaptiverly altered according to channel state. The decoder permits 4-step pipelined processing : (1) syndrome calculation (2) MEA block (3) error magnitude calculation (4) decoder failure check. Each step is designed to form a structure suitable for decoding a RS code with varying block length. A new architecture is proposed for a MEA block in step (2) and an architecture of outputting in reversed order is employed for a polynomial evaluation in step (3). To maintain to throughput rate with less circuitry, the MEA block uses not only a multiplexing and recursive technique but also an overclocking technique. The adaptive RS decoder over GF($2^8$) with the maximal error correcting capability of 10 has been designed in VHDL, and successfully synthesized in a FPGA.