• Journal of the Institute of Electronics Engineers of Korea TC
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• v.43 no.7 s.349
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• pp.143-149
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• 2006

Turbo code, a kind of error correction coding technique, has been used in the field of digital mobile communication system. As the number of iterations increases, it can achieves remarkable BER performance over AWGN channel environment. However, if the number of iterations is increased in the several channel environments, any further iteration results in very little improvement, and requires much delay and computation in proportion to the number of iterations. To solve this problems, it is necessary to device an efficient criterion to stop the iteration process and prevent unnecessary delay and computation. In this paper, it proposes an efficient and simple criterion for stopping the iteration process in turbo decoding. By using sign changed ratio of extrinsic information values in turbo decoder, the proposed algorithm can largely reduce the average number of iterations without BER performance degradation. As a result of simulations, the average number of iterations is reduced by about $12.48%{\sim}22.22%$ compared to CE algorithm and about $20.43%{\sim}54.02%$ compared to SDR algorithm.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.42 no.12
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• pp.127-134
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• 2005

Lost packet robustness is an most important quality measure for voice over IP networks(VoIP). Recovery of the lost packet from the received information is crucial to realize this robustness. So, this paper proposes the lost packet recovery method from the received information for real-time communication for CELP coder. The proposed BS-PLC (Both Side Packet Loss Concealment) based WSOLA(Waveform Shift OverLab Add) allow the lost packet to be recovered from both the 'previous' and 'next' good packet as the LP parameter and the excitation signal are respectively recovered. The burst of packet loss is modeled by Gilbert model. The proposed scheme is applied to G.729 most used in VoIP and is evaluated through the SNR(signal to noise) and the MOS(Mean Opinion Score) test. As a simulation result, The proposed scheme provide 0.3 higher in Mean Opinion Score and 2 dB higher in terms of SNR than an error concealment procedure in the decoder of G.729 at $20\%$ average packet loss rate.

• Phonetics and Speech Sciences
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• v.12 no.2
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• pp.29-37
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• 2020

Recently, the neural network-based deep learning algorithm has dramatically improved performance compared to the classical Gaussian mixture model based hidden Markov model (GMM-HMM) automatic speech recognition (ASR) system. In addition, researches on end-to-end (E2E) speech recognition systems integrating language modeling and decoding processes have been actively conducted to better utilize the advantages of deep learning techniques. In general, E2E ASR systems consist of multiple layers of encoder-decoder structure with attention. Therefore, E2E ASR systems require data with a large amount of speech-text paired data in order to achieve good performance. Obtaining speech-text paired data requires a lot of human labor and time, and is a high barrier to building E2E ASR system. Therefore, there are previous studies that improve the performance of E2E ASR system using relatively small amount of speech-text paired data, but most studies have been conducted by using only speech-only data or text-only data. In this study, we proposed a semi-supervised training method that enables E2E ASR system to perform well in corpus in different domains by using both speech or text only data. The proposed method works effectively by adapting to different domains, showing good performance in the target domain and not degrading much in the source domain.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.13 no.8
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• pp.804-810
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• 2002

The code division multiple access(CDMA) system capacity is limited by the amount of interference of the system. To reduce the unnecessary interference, this paper proposes optimized cell identification codes for site selection diversity transmission(SSDT) power control in wideband code division multiple access system of third generation partnership project(3GPP). The main objective of SSDT power control is to transmit on the downlink from the primary cell, and thus reducing the interference caused by the multiple transmission. In order to select a primary cell, each cell is assigned a temporary identification(ID) and user equipment(UE) periodically informs a primary cell ID to the connecting cells during soft handover. The non-primary cells selected by UE do not transmit the dedicated physical data channel(DPDCH) to reduce the interference. A major issue with the SSDT technology is the impact of uplink symbol errors on its performance. These errors can corrupt the primary ID code and this may lead to wrong decoding in the base station receivers. The proposed SSDT cell ID codes are designed to minimize the problem and to be easily decoded using simple fast Hadamard transformation(FHT) decoder.

• Journal of Broadcast Engineering
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• v.9 no.4 s.25
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• pp.391-401
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• 2004

In this paper, we introduce a novel coding method to efficiently enable spatial random access for high resolution video. In terms of resolution and display size, standard display devices (such as cathode-ray tubes. monitors. PDAs, and LCDs) do not sufficiently support high resolution video such as digital cinema and panoramic video. Currently, users have no choice but to view video at lower resolution as a result of down-sampling, or only a partial region of the video due to display size limitations. Our proposed method. which we call the B-map, represents the set of starting locations of the coded segments in a picture frame. This information, or B-map, is first sent to the decoder prior to the coded data stream of the frame and is then used for fast local decoding. To test our method, we compare our B-map with JPEG tiling and the JPEG Resynchronization marker. Experimental results show that the proposed coding method requires less overhead than existing methods during the same decoding time. The results show promise for future panoramic or digital cinema applications.

• Journal of Broadcast Engineering
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• v.10 no.3
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• pp.413-421
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• 2005

As the population of HDTV is growing, among many useful features of modern set top boxes (STBs) or digital video recorders (DVRs), video browsing, visual bookmark, and picture-in-picture capabilities are very frequently required. These features typically employ reduced-size versions of video frames, or thumbnail images. Most thumbnail generation approaches generate DC images directly from a compressed video stream. A discrete cosine transform (DCT) coefficient for which the frequency is zero in both dimensions in a compressed block is called a DC coefficient and is simply used to construct a DC image. If a block has been encoded with field DCT, a few AC coefficients are needed to generate the DC image in addition to a DC coefficient. However, the bit length of a codeword coded with variable length coding (VLC) cannot be determined until the previous VLC codeword has been decoded, thus it is required that all codewords should be fully decoded regardless of their necessary for DC image generation. In this paper, we propose a method especially for fast DC image generation from an I-frame using multiple-symbol lookup table (mLUT). The experimental results show that the method using the mLUT improves the performance greatly by reducing LUT count by 50$\%$.

• Journal of Broadcast Engineering
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• v.15 no.1
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• pp.52-62
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• 2010

Recently, DVC (Distributed Video Coding) is attracting a lot of research works since this enables us to implement a light-weight video encoder by distributing the high complex tasks such as motion estimation into the decoder side. In order to improve the coding efficiency of the DVC, the existing works have been focused on the efficient generation of side information (SI) or the virtual channel modeling which can describe the statistical channel noise well. But, in order to improve the overall performance, this paper proposes a new scheme that can be implemented with simple bit operations without introducing complex operation. That is, the performance of the proposed scheme is enhanced by using the fact that the Wyner-Ziv (WZ) frame and side information are highly correlated, and by reducing the effect of virtual channel noise which tends to be clustered in some regions. For this aim, this paper proposes an efficient pixel-domain WZ (PDWZ) CODEC which effectively exploits the statistical redundancy by using the code conversion and Gray code, and then reduces the channel noise by using the bit interleaver. Through computer simulations, it is shown that the proposed scheme can improve the performance up to 0.5 dB in objective visual quality.

• Journal of Broadcast Engineering
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• v.13 no.2
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• pp.188-199
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• 2008

In conventional video coding, encoder complexity is much higher than that of decoder. However, investigations for lightweight encoder to eliminate motion prediction/compensation claiming most complexity in encoder have recently become an important issue. The Wyner-Ziv coding is one of the representative schemes for the problem and, in this scheme, since encoder generates only parity bits of a current frame without performing any type of processes extracting correlation information between frames, it has an extremely simple structure compared to conventional coding techniques. However, in Wyner-Ziv coding, channel decoding errors occur when noisy side information is used in channel decoding process. These channel decoding errors appear more frequently, especially, when there is not enough correlation between frames to generate accurate side information and, as a result, those errors look like Salt & Pepper type noise in the reconstructed frame. Since this noise severely deteriorates subjective video quality even though such noise rarely occurs, previously we proposed a computationally extremely light encoding method based on selective median filter that corrects such noise using spatial correlation of a frame. However, in the previous method, there is a problem that loss of texture from filtering may exceed gain from error correction by the filter for video sequences having complex torture. Therefore, in this paper, we propose an improved lightweight encoding method that minimizes loss of texture detail from filtering by allowing information of texture and that of noise in side information to be utilized by the selective median filter. Our experiments have verified average PSNR gain of up to 0.84dB compared to the previous method.

• The Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.13 no.5
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• pp.338-343
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• 2020

In this paper, it was shown that the decoding method of Hamming codes using soft values can be applied to molecular communication channels. A soft value criterion that can be used for decoding of Hamming codes for a molecular communication channel was proposed, and it has been shown through simulation that the decoding method using these values can improve reliability even in the molecular communication channel. A diffusion-based molecular communication channel was assumed, and information symbols were transmitted using BCSK modulation. After demodulating the number of molecules absorbed by the receiver at each symbol interval with an appropriate threshold, the number of molecules is no longer used. In this paper, the BER performance of the decoder was improved by utilizing information on the number of molecules that are no longer used as soft values in the decoding process. Simulation was performed to confirm the improvement in BER performance. When the number of molecules per bit is 600, the error rate of the Hamming code (15,11) was improved about 5.0×10^-3 to the error rate of the BCSK system without the Hamming code. It can be seen that the error rate of (15,11) Hamming code with the soft values was improved to the same extent. In the case of (7,4) Hamming code, the result is similar to that of (15,11) Hamming code. Therefore, it can be seen that the BER performance of the Hamming code can be greatly improved even in the molecular communication channel by using the difference between the number of molecules absorbed by the receiver and the threshold value as a soft value.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.25 no.9A
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• pp.1290-1298
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• 2000

Turbo Code decoder is an iterate decoding technology, which extracts extrinsic information from the bit to be decoded by calculating both forward and backward metrics, and uses the information to the next decoding step Turbo Code shows excellent performance, approaching Shannon Limit at the view of BER, when the size of Interleaver is big and iterate decoding is run enough. But it has the problems which are increased complexity and delay and difficulty of real-time processing due to Interleaver and iterate decoding. In this paper, it is analyzed that MAP(maximum a posteriori) algorithm which is used as one of Turbo Code decoding, and the factor which determines its performance. MAP algorithm proceeds iterate decoding by determining soft decision value through the environment and transition probability between all adjacent bits and received symbols. Therefore, to improve the performance of MAP algorithm, the trust between adjacent received symbols must be ensured. However, MAP algorithm itself, can not do any action for ensuring so the conclusion is that it is needed more algorithm, so to decrease iterate decoding. Consequently, MAP algorithm and Turbo Code performance are analyzed in the nongaussian channel applying Robust equalization technique in order to input more trusted information into MAP algorithm for the received symbols.