• IEIE Transactions on Smart Processing and Computing
• /
• v.5 no.4
• /
• pp.243-249
• /
• 2016

This paper proposes a new watermarking algorithm to embed watermarks in thr process of sample adaptive offsets (SAO) for high efficiency video coding (HEVC) compressed videos. The proposed method embeds two-bit watermark into the SAO offsets for each coding tree unit (CTU). To minimize visual quality degradation caused by embedding watermark, watermark bits are embedded into SAO offset depending on the SAO types of block. Furthermore, the embedded watermark can be extracted by simply adding four offsets and checking their least significant bits (LSB) at the decoder side. The experimental results show that the proposed method achieves 0.3% BD-rate increase without much visual quality degradation. Two-bit watermark for each CTU is embedded for more bit watermarking. In addition, the proposed method requires negligible computational load for watermark insertion and extraction.

• Proceedings of the KIEE Conference
• /
• 2007.04a
• /
• pp.292-294
• /
• 2007

Recently, for satisfying many application demands such as coding delay, computing power, transporting channel characteristic, etc, many profiles are supported in video coding standards. Therefore, in transcoding between same standards or between other standards, the functional difference of profiles supported by application occur many problems. In this paper, transcoding MPEG-2 main profile to H.264/AVC baseline profile which has restriction in the number of reference frame is focused. In this case, the bidirectional prediction supported in MPEG-2 main profile is not supported in H.264/AVC baseline profile. Also, in the restriction of reference frame, motion vectors in the MPEG-2 decoder as predictor should be adjusted. In this paper, the proposed algorithm is based on the characteristic of which motion. vector is uniform according to the distance from reference frame. The adaptive search techniques through the determination of the uniformity extremely reduce the computational complexity.

• Journal of Communications and Networks
• /
• v.13 no.3
• /
• pp.257-265
• /
• 2011

In this paper, a cooperative relaying protocol called soft-decision-and-forward (SDF) with multiple antennas in each node is introduced. SDF protocol exploits the soft decision source symbol values from the received signal at the relay node. For orthogonal transmission (OT), orthogonal codes including Alamouti code are used and for non-orthogonal transmission (NT), distributed space-time codes are designed by using a quasi-orthogonal space-time block code. The optimal maximum likelihood (ML) decoders for the proposed protocol with low decoding complexity are proposed. For OT, the ML decoders are derived as symbolwise decoders while for NT, the ML decoders are derived as pairwise decoders. It can be seen through simulations that SDF protocol outperforms AF protocol for both OT and NT.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.31 no.10A
• /
• pp.976-981
• /
• 2006

In this paper, we describe a hybrid ZF-SD method. The method is based on dimensionality reduction via predecoding and cancellation of those symbols that can be quickly and reliably detected by a linear decoder. The proposed method shows BER performance similar to SD but with much lower computational complexity than SD.

• ETRI Journal
• /
• v.28 no.5
• /
• pp.664-667
• /
• 2006

We propose an efficient audio transcoding algorithm that can convert audio streams from terrestrial digital television broadcasting service stations to those for terrestrial digital multimedia broadcasting hand-held receivers. The proposed algorithm avoids the complicated psychoacoustic analysis by calculating the scalefactors of the bit-sliced arithmetic coding encoder directly from the signal-to-noise ratio parameters of the AC-3 decoder. The bit-allocation process is also simplified by cascading the nested distortion control loop. Through subjective evaluation, it is shown that the proposed algorithm provides comparable audio quality to tandem coding but it requires much smaller complexity.

• ETRI Journal
• /
• v.29 no.1
• /
• pp.103-106
• /
• 2007

Pilot-based coding (PBC), which is used for lossless bit rate reduction of audio coding, has been recently proposed for MPEG Surround. We propose extended PBC for further lossless bit rate reduction of MPEG Surround. Extended PBC selects the number of pilots depending on the parameter band number and the type of spatial parameter. It then encodes the pilots and the relevant difference data. Experiments show that extended PBC is more effective than the original PBC, especially for high bit rate modes, with a negligible complexity increase on the decoder side.

• Proceedings of the Korean Society of Broadcast Engineers Conference
• /
• 2003.11a
• /
• pp.7-10
• /
• 2003

인터프레임 웨이블렛 부호화(Interframe Wavelet Coding)는 3D 서브밴드(Subband) 부호화라고도 하며, 기존의 DCT기반 Hybrid 동영상 부호화 방식에 비해 압축 효율이 우수하고. 특히 스케일러빌리티 기능이 뛰어난 부호화 방법이다. 인터프레임 웨이블렛 부호화 방법에서 복호화 과정 중 가장 연산 량이 많이 요구되는 역(inverse) 웨이블렛 변환이다 역 웨이블렛 변환의 연산 량은 복호화 과정에서 적용된 웨이블렛 변환과 동일한 연산량을 요구한다. 이는 순방향과 역방향에서 동일 길이의 필터와 분해 레벨을 사용해야 하기 때문이다. 이 웨이블렛 변환의 연산 량을 줄이기 위해 본 논문에서는 기존의 시간 밴드 영상에 대해 동일 한 웨이블렛 필터를 사용하여 공간 웨이블렛 필터를 적용하던 것을. 로우밴드에는 9/7 필터를 적용하고 하이 밴드에는 Haar필터를 사용하는 방법을 제안한다. PSNR 실험에서 기존의 9/7 필터만을 사용하는 경우와 비교한 결과 거의 차이가 없었다.

• KSII Transactions on Internet and Information Systems (TIIS)
• /
• v.15 no.3
• /
• pp.1156-1173
• /
• 2021

In this paper, we propose a new joint precoder and postcoder design strategy to support multiple streams per user in multiuser multiple-input multiple-output (MIMO) systems. We propose two step precoding strategies using equal channel gain decomposition and block diagonalization at the transmitter. With the proposed precoder, the multiuser MIMO channel can be decomposed into multiple parallel channels with equal channel gain per user. After applying receive postcoder which is generated and sent by the transmitter, we can use ML based decoder per stream to achieve full receive diversity. Achievable sum rate bound and diversity performance of the proposed algorithm are presented with feedback signaling design and quantitative complexity analysis. Simulation results show that the proposed algorithm asymptotically approaches to the sum rate capacity of the MIMO broadcast channel while maintaining full diversity order.

• ETRI Journal
• /
• v.45 no.2
• /
• pp.277-290
• /
• 2023

This paper proposes a method that can reduce the complexity of a system matrix by analyzing the characteristics of a pseudoinverse matrix to receive a binomial frequency division multiplexing (BFDM) signal and decode it using the least squares (LS) method. The system matrix of BFDM can be expressed as a band matrix, and as this matrix contains many zeros, its amount of calculation when generating a transmission signal is quite small. The LS solution can be obtained by multiplying the received signal by the pseudoinverse matrix of the system matrix. The singular value decomposition of the system matrix indicates that the pseudoinverse matrix is a band matrix. The signal-to-interference ratio is obtained from their eigenvalues. Meanwhile, entries that do not contribute to signal generation are erased to enhance calculation efficiency. We decode the received signal using the pseudoinverse matrix and the removed pseudoinverse matrix to obtain the bit error rate performance and to analyze the difference.

• The Journal of the Acoustical Society of Korea
• /
• v.30 no.4
• /
• pp.190-196
• /
• 2011

Bandwidth extension is a technique to improve speech quality and intelligibility, extending from 300-3400 Hz narrowband speech to 50-7000 Hz wideband speech. This paper designs an artificial bandwidth extension (ABE) module embedded in the AMR (adaptive multi-rate) decoder, reducing LPC/LSP analysis and algorithm delay of the ABE module. We also introduce a fast search codebook mapping method for ABE, and design a low power BWE technique based on the AMR decoder. The proposed ABE method reduces the computational complexity and the algorithm delay, respectively, by 28 % and 20 msec, compared to the traditional DTE (decode then extend) method. We also introduce a weighted classified codebook mapping method for constructing the spectral envelope of the wideband speech signal.