• KSII Transactions on Internet and Information Systems (TIIS)
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• v.17 no.3
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• pp.999-1021
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• 2023

Differential spatial modulation uses the antenna index to transmit information, which improves the spectral efficiency, and completely bypasses any channel side information in the recommended setting. A generalized distributed multiple turbo coded-cooperative differential spatial modulation based on distributed multiple turbo code is put forward and its performances in Rayleigh fading channels is analyzed. The generalized distributed multiple turbo coded-cooperative differential spatial modulation scheme is a coded-cooperation communication scheme, in which we proposed a new joint parallel iterative decoding method. Moreover, the code matched interleaver is considered to be the best choice for the generalized multiple turbo coded-cooperative differential spatial modulation schemes, which is the key factor of turbo code. Monte Carlo simulated results show that the proposed cooperative differential spatial modulation scheme is better than the corresponding non-cooperative scheme over Rayleigh fading channels in multiple input and output communication system under the same conditions. In addition, the simulation results show that the code matched interleaver scheme gets a better diversity gain as compared to the random interleaver.

• Proceedings of the IEEK Conference
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• 2000.09a
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• pp.41-46
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• 2000

In this paper, we analyze the performance of W-CDMA systems using turbo code in the wideband multipath channel. The strength of each multipath component is assumed to be Rayleigh distributed. Turbo codes of long block sizes have been known to show very good performance in a AWGN channel. Recently, turbo codes of short block sizes suitable for real time communication systems have attracted a lot of attention. Thus, in this paper we consider the turbo code of 1/3 code rate and short frame size of 192 bits in wideband mobile multipath channel. We analyze the performance of W-CDMA systems with 10MHz bandwidths in the case of employing RAKE receiver with not only MRC diversity but also turbo code.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.36 no.2C
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• pp.57-64
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• 2011

A two-way relay channel is a bidirectional cooperative communication channel between two nodes using a relay. In many cooperative communication schemes, a relay transmits its data to each node using separate channels. However, in the two-way relay channel, a relay can broadcast the network-coded signal to both nodes in a same time slot, which can increase the system throughput. In this paper, a new cooperative network error correcting scheme using distributed turbo code in a two-way relay channel is proposed. The proposed scheme not only increases the system throughput using network code but also improves the performance by utilizing the LLR information from relay node and other user node through distributed turbo code. Also, a power allocation scheme is investigated for various channel conditions to improve the system performance.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.37 no.1A
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• pp.34-42
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• 2012

Low density parity check (LDPC) code is widely used, since it shows superior performance close to Shannon limit and its decoding complexity is lower than turbo code. Recently, it is used as a channel code to decode Wyner-Ziv frames in distributed video coding (DVC) system. In this paper, we propose an efficient method to design the parity check matrix H of LDPC codes. In order to apply LDPC code to DVC system, the LDPC code should have rate compatibility. Thus, we also propose a method to merge check nodes of LDPC code to attain the rate compatibility. LDPC code is designed using ACE algorithm and check nodes are merged for a given code rate to maximize the error correction capability. The performance of the designed LDPC code is analyzed extensively by computer simulations.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.36 no.6C
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• pp.368-375
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• 2011

In the distributed video coding system, the parity bits, which are generated in encoders, are used to reconstruct Wyner-Ziv frames. Since the original Wyner-Ziv frames are not known in decoders, the efficient correlation noise modeling for turbo or LDPC code is necessary. In this paper, an efficient correlation noise modeling method is proposed and the performance is analyzed. The method to estimate the quantization parameters for key frames, which are encoded using H.264 intraframe coding technique, is also proposed. The performance of the proposed system is evaluated by extensive computer simulations.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.47 no.6
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• pp.66-74
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• 2010

Recently distributed video coding (DVC) is spotlighted for the environment which has restriction in computing resource at encoder. Wyner-Ziv (WZ) coding is a representative scheme of DVC. The WZ encoder independently encodes key frame and WZ frame respectively by conventional intra coding and channel code. WZ decoder generates side information from reconstructed two key frames (t-1, t+1) based on temporal correlation. The side information is regarded as a noisy version of original WZ frame. Virtual channel noise can be removed by channel decoding process. So the performance of WZ coding greatly depends on the performance of channel code. Among existing channel codes, Turbo code and LDPC code have the most powerful error correction capability. These channel codes use stochastically iterative decoding process. However the iterative decoding process is quite time-consuming, so complexity of WZ decoder is considerably increased. Analysis of the complexity of LPDCA with real video data shows that the portion of complexity of LDPCA decoding is higher than 60% in total WZ decoding complexity. Using the HDA (Hard Decision Aided) method proposed in channel code area, channel decoding complexity can be much reduced. But considerable RD performance loss is possible according to different thresholds and its proper value is different for each sequence. In this paper, we propose an adaptive HDA method which sets up a proper threshold according to sequence. The proposed method shows about 62% and 32% of time saving, respectively in LDPCA and WZ decoding process, while RD performance is not that decreased.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2011.07a
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• pp.284-286
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• 2011

LDPC(low density parity check) 부호는 낮은 복잡성과 Shannon의 한계에 근접하는 오류 정정 능력을 보이기 때문에 turbo 부호와 함께 많은 응용분야에 적용되고 있다. 본 논문에서는 분산 동영상 부호화(distributed video coding: DVC) 시스템을 위한 부호율 적응적인(rate adaptive) LDPC 부호를 설계하기 위하여 패리티 점검 노드를 병합하는 방법을 제안한다. ACE(approximation cycles EMD) 알고리즘을 기반으로 효율적인 LDPC 부호를 설계하고 부호율 적응적인 특성을 갖기 위해 일정한 범위를 지정하고 지정된 범위에 따라 패리티 점검 노드를 병합한다. 그리고 ACE 알고리즘의 계수와 degree distribution을 변화시키면서 성능을 해석한다.

• Journal of Broadcast Engineering
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• v.14 no.2
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• pp.144-153
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• 2009

Recently, as the necessity of a light-weighted video encoding technique has been rising for applications such as UCC(User Created Contents) or Multiview Video, Distributed Video Coding(DVC) where a decoder, not an encoder, performs the motion estimation/compensation taking most of computational complexity has been vigorously investigated. Wyner-Ziv coding reconstructs an image by eliminating the noise on side information which is decoder-side prediction of original image using channel code. Generally the side information of Wyner-Ziv coding is generated by using frame interpolation between key frames. The channel code such as Turbo code or LDPC code which shows a performance close to the Shannon's limit is employed. The noise model of Wyner-Ziv coding for channel decoding is called Virtual Channel Noise and is generally modeled by Laplacian or Gaussian distribution. In this paper, we propose a Wyner-Ziv coding method based on the frequency-adaptive channel noise modeling in transform domain. The experimental results with various sequences prove that the proposed method makes the channel noise model more accurate compared to the conventional scheme, resulting in improvement of the rate-distortion performance by up to 0.52dB.

• 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.