• Journal of Broadcast Engineering
• /
• v.23 no.6
• /
• pp.768-779
• /
• 2018

In this paper, we propose a novel video quality function for video processing based on MSSSIM to select an appropriate video processing filter and to accommodate multiple processing filters to each pixel block in a picture frame by a mathematical selection law so as to maintain video quality and to reduce the bitrate of compressed video. In viewpoint of video compression, since the properties of video quality and bitrate is different for each picture of video frames and for each areas in the same frame, it is difficult for the video filter with single property to satisfy the object of increasing video quality and decreasing bitrate. Consequently, to maintain the subjective video quality in spite of decreasing bitrate, we propose the methodology about the MSSSIM as the measure of subjective video quality, the KL-Divergence as the measure of bitrate, and the combination method of those two measurements. Moreover, using the proposed combinatorial measurement, when we use the multiple image filters with mutually different properties as a pre-processing filter for video, we can verify that it is possible to compress video with maintaining the video quality under decreasing the bitrate, as possible.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.32 no.4C
• /
• pp.411-417
• /
• 2007

The rate control is an essential component in video coding to provide better quality under given coding constraints, such as channel capacity, frame rates, etc. In general, source data cannot be described as a single distribution in a video coding, hence it can cause an exhaustive approximation problem. It drops a coding efficiency under weak channel environments, such as mobile communications. In this paper, we design a new quantization parameter decision model that is based on a rate-distortion function of generalized Gaussian distribution. In order to adaptively express various source data distribution, we decide a shape parameter by observing a ratio of samples, which have a small value. For experiment, the proposed algorithm is implemented into H.264/AVC video codec, and its performance is compared with that of MPEG-2 TM5, H.263 TMN8 rate control algorithm. As shown in simulation results, the proposed algorithm provides an improved quality rather than previous algorithms and generates the number of bits closed to the target bits.

• Journal of the Institute of Electronics Engineers of Korea TC
• /
• v.43 no.3 s.345
• /
• pp.27-31
• /
• 2006

Millimeter over Fiber (MMoF) technique modulates millimeter wave signal optically to transmit it through an optical fiber for long distances with small loss. MMoF system usually uses optical single sideband (OSSB) modulation scheme to reduce fiber chromatic dispersion and obtain high bandwidth efficiency. The optical link of MMoF system using OSSB is treated as a nonlinear amplifier, and the AM/AM characteristic function of the amplifier is a Bessel function of the first kind of order 1. In this paper, we investigate the performance of OFDM MMoF system considering nonlinearity of OSSB modulation. We estimate a power of the nonlinear distortion noise to analyze the theoretical bit error rate(BER), and perform a simulation to verify the theoretical BER.

• Journal of the Institute of Electronics Engineers of Korea SP
• /
• v.44 no.5
• /
• pp.73-83
• /
• 2007

In this paper, we propose macroblock(MB)-based adaptive interpolation filter method for improving coding efficiency in H.264/AVC. In the proposed method, nine separable two-dimensional(2D) interpolation filters are applied for precisely compensating motions in various directions. The optimal cost function which considers the bit rate and distortion for coding the MB is defined. The filter is adaptively selected per MB for minimizing the defined cost function. In the experimental results, the proposed method shows more excellent in coding efficiency than the conventional methods for the various standard $QCIF(176{\times}144)/CIF(352{\times}288)$ video test sequences. It leads to about 6.25%(1 reference frame) and 3.46%(5 reference frames) bit rate reduction on average compared to the H.264/AVC.

• The Journal of the Acoustical Society of Korea
• /
• v.24 no.7
• /
• pp.423-430
• /
• 2005

In this paper. a microphone array beamforming algorithm that reduces the signal distortion caused by reverberation and near-field effect in car environment is proposed. When reverberation or near-field effect is present, an optimum beamformer should be constructed with a steering vector consisting of transfer functions between source and microphones, but it is generally difficult to estimate transfer functions on-line without knowledge of the source signal. Instead, a sub-optimal beamforming algorithm that reduces signal distortion is proposed. It is constructed with steering vectors consisting of relative transfer functions between reference sensor and other sensors. In order to evaluate the performance of the proposed algorithm. we had recorded noisy speech database in a car, and performed speech recognition experiments with HMM Toolkit (HTK) released by Cambridge University. The recognition rate of the proposed algorithm was 15 percents higher than that of the conventional far-field beamformers in best case.

• Journal of Advanced Navigation Technology
• /
• v.16 no.2
• /
• pp.234-239
• /
• 2012

Next generation mobile communication systems have been studied and applied to support various services. In next generation mobile communication systems, the most interested research is the integration of various communication systems and the offer of various services by using high-speed data transmission. The integration of communication systems has been researched by using multi standard modem, while the high-speed data transmission for the offer of various services has been applied by using OFDM. This paper has proposed the method to reduce PAPR by using multi standard modem. with EVM, this paper has also suggested the method to compensate in-band distortion while reducing PAPR. By using the proposed methods, this paper has analyzed and simulated the decrease efficiency of PAPR, the performance of CCDF, and the performance of BER in next generation mobile communication systems. The simulation results improved the performance of next generation mobile communication system can be seen that.

• Journal of the Korea Academia-Industrial cooperation Society
• /
• v.22 no.3
• /
• pp.627-632
• /
• 2021

ITL (information-theoretic learning) has been applied successfully to adaptive signal processing and machine learning applications, but there are difficulties in deciding the kernel size, which has a great impact on the system performance. The correntropy algorithm, one of the ITL methods, has superior properties of impulsive-noise robustness and channel-distortion compensation. On the other hand, it is also sensitive to the kernel sizes that can lead to system instability. In this paper, considering the sensitivity of the kernel size cubed in the denominator of the cost function slope, a new adaptive kernel estimation method using the rate of change in error power in respect to the kernel size variation is proposed for the correntropy algorithm. In a distortion-compensation experiment for impulsive-noise and multipath-distorted channel, the performance of the proposed kernel-adjusted correntropy algorithm was examined. The proposed method shows a two times faster convergence speed than the conventional algorithm with a fixed kernel size. In addition, the proposed algorithm converged appropriately for kernel sizes ranging from 2.0 to 6.0. Hence, the proposed method has a wide acceptable margin of initial kernel sizes.

• Journal of Broadcast Engineering
• /
• v.11 no.1 s.30
• /
• pp.116-129
• /
• 2006

The progress of data transmission technology through the Internet has spread a variety of realistic contents. One of such contents is multi-view video that is acquired from multiple camera sensors. In general, the multi-view video processing requires encoders and decoders as many as the number of cameras, and thus the processing complexity results in difficulties of practical implementation. To solve for this problem, this paper considers a simple multi-view system utilizing a single encoder and a single decoder. In the encoder side, input multi-view YUV sequences are combined on GOP units by a video mixer. Then, the mixed sequence is compressed by a single H.264/AVC encoder. The decoding is composed of a single decoder and a scheduler controling the decoding process. The goal of the scheduler is to assign approximately identical number of decoded frames to each view sequence by estimating the decoder utilization of a Gap and subsequently applying frame skip algorithms. Furthermore, in the frame skip, efficient frame selection algorithms are studied for H.264/AVC baseline and main profiles based upon a cost function that is related to perceived video quality. Our proposed method has been performed on various multi-view test sequences adopted by MPEG 3DAV. Experimental results show that approximately identical decoder utilization is achieved for each view sequence so that each view sequence is fairly displayed. As well, the performance of the proposed method is examined in terms of bit-rate and PSNR using a rate-distortion curve.

• Journal of the Korean Society for Aeronautical & Space Sciences
• /
• v.50 no.11
• /
• pp.801-807
• /
• 2022

In this paper, we propose an ultra compact S-Band PCM/FM telemetry transmitter. The equipment is compact, so it can be applied to a limited space and capable of stable data transmission was designed and manufactured even with specifications set differently for each operating environment and system. RF direct conversion structure is used for the miniaturization of equipment, an RF transmission board, Power distribution board, and a signal processing board were implemented on a single PCB, so that the function of the transmitter could be performed with a minimum device. According to the target specification, variable output of 1~10W and variable data rate of 390kbps~12.5Mbps is possible in S-Band(2,200~2,400MHz) without degradation of performance. To verify the performance of the equipment, the RF performance test and BER measurement test were performed after the equipment was manufactured. It was confirmed that the OBW, null-to-null bandwidth, 1st IMD, Spurious emission, Phase noise specification of the PCM/FM modulated signal which is presented by the IRIG standard were satisfied, and we can confirm the data received using the transmitter inspection equipment were transmitted normally without distortion.

• KSII Transactions on Internet and Information Systems (TIIS)
• /
• v.10 no.3
• /
• pp.1195-1211
• /
• 2016

The emerging video coding standard High Efficiency Video Coding (HEVC) has shown almost 40% bit-rate reduction over the state-of-the-art Advanced Video Coding (AVC) standard but at about 40% computational complexity overhead. The main reason for HEVC computational complexity is the inter prediction that accounts for 60%-70% of the whole encoding time. In this paper, we propose several fast coding unit (CU) encoding schemes based on the Merge mode and motion estimation information to reduce the computational complexity caused by the HEVC inter prediction. Firstly, an early Merge mode decision method based on motion estimation (EMD) is proposed for each CU size. Then, a Merge mode based early termination method (MET) is developed to determine the CU size at an early stage. To provide a better balance between computational complexity and coding efficiency, several fast CU encoding schemes are surveyed according to the rate-distortion-complexity characteristics of EMD and MET methods as a function of CU sizes. These fast CU encoding schemes can be seamlessly incorporated in the existing control structures of the HEVC encoder without limiting its potential parallelization and hardware acceleration. Experimental results demonstrate that the proposed schemes achieve 19%-46% computational complexity reduction over the HEVC test model reference software, HM 16.4, at a cost of 0.2%-2.4% bit-rate increases under the random access coding configuration. The respective values under the low-delay B coding configuration are 17%-43% and 0.1%-1.2%.