• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.9C
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• pp.877-886
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• 2005

Various frame recovery algorithms have been suggested to overcome the communication quality degradation problem due to Internet-typical impairments on Voice over IP(VoIP) communications. In this paper, we propose a new receiver-based recovery method which is able to enhance recovered speech quality with almost free computational cost and without an additional increment of delay and bandwidth consumption. Most conventional recovery algorithms try to recover the lost or erroneous speech frames by reconstructing missing coefficients or speech signal during speech decoding process. Thus they eventually need to modify the decoder software. The proposed frame recovery algorithm tries to reconstruct the missing frame itself, and does not require the computational burden of modifying the decoder. In the proposed scheme, the Vector Quantization(VQ) codebook indices of the erased frame are directly estimated by referring the pre-computed VQ Codebook Index Interpolation Tables(VCIIT) using the VQ indices from the adjacent(previous and next) frames. We applied the proposed scheme to the ITU-T G.723.1 speech coder and found that it improved reconstructed speech quality and outperforms conventional G.723.1 loss recovery algorithm. Moreover, the suggested simple scheme can be easily applicable to practical VoIP systems because it requires a very small amount of additional computational cost and memory space.

• ETRI Journal
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• v.25 no.5
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• pp.356-368
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• 2003

This paper analyzes the characteristics of packet losses in mobile computing environments based on the Gilbert model and then describes a mechanism that can recover the lost audio packets using redundant data. Using information periodically reported by a receiver, the sender dynamically adjusts the amount and offset values of redundant data with the constraint of minimizing the bandwidth consumption of wireless links. Since mobile computing environments can be often characterized by frequent and consecutive packet losses, loss recovery mechanism need to deal efficiently with both random and consecutive packet losses. To achieve this, the suggested mechanism uses relatively large, discontinuous exponential offset values. That gives the same effect as using both the sequential and interleaving redundant information. To verify the effectiveness of the mechanism, we extended and implemented RTP/RTCP and applications. The experimental results show that our mechanism, with an exponential offset, achieves a remarkably low complete packet loss rate and adapts dynamically to the fluctuation of the packet loss pattern in mobile computing environments.

• Journal of Positioning, Navigation, and Timing
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• v.5 no.3
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• pp.109-118
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• 2016

The present paper proposes a method that can estimate a roll rate of spinning vehicles utilizing GPS receivers. The proposed method analyzes a relation between received signal and correlation value and utilizes a phenomenon that received signal power that changes according to a signal incident direction affects a correlation value. That is, a roll-rate estimation method using zero crossing detection method for correlation value, which has sinusoidal periodicity according to rotations of vehicles, is proposed. A correlation value in real environments experiences a jitter so that the proposed method includes a pre-processing filter and detection threshold setting way is also considered to reduce the effect of received signal power. In order to verify the operation of the proposed method and analyze the performance, a signal generator and software-defined receiver (SDR) are designed. The signal generator generates intermediate frequency (IF) signal by taking the rotation of vehicles, antenna gain, and signal power into consideration, and a correlation value is acquired by taking the generated IF signals into consideration. Using the generated correlation value, the operation of the proposed roll rate estimation method is verified and the performance is analyzed.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.9 no.12
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• pp.5170-5188
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• 2015

This paper proposes a novel progressive secret image-hiding scheme based on the inpainting technique, the vector quantization technique (VQ) and the exploiting modification direction (EMD) technique. The proposed scheme first divides the secret image into non-overlapping blocks and categorizes the blocks into two groups: complex and smooth. The blocks in the complex group are compressed by VQ with PCA sorted codebook to obtain the VQ index table. Instead of embedding the original secret image, the proposed method progressively embeds the VQ index table into the cover images by using the EMD technique. After the receiver recovers the complex parts of the secret image by decoding the VQ index table from the shadow images, the smooth parts can be reconstructed by using the inpainting technique based on the content of the complex parts. The experimental results demonstrate that the proposed scheme not only has the advantage of progressive data hiding, which involves more shadow images joining to recover the secret image so as to produce a higher quality steganography image, but also can achieve high hiding capacity with acceptable recovered image quality.

• Ocean and Polar Research
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• v.31 no.1
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• pp.1-9
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• 2009

When evaluating acoustic methods for measuring physical parameters in the ocean, economical and technical considerations are paramount. As an indirect method of estimating ocean dynamics, acoustic tomography has advantages over more conventional approaches. It allows the reconstruction of temperature and flow fields from the acoustic impulse time-of-flight measured along the rays propagating from the source to the receiver. However, many problems require complicated and expensive systems. To use the acoustic tomography method to best effect, developing hardware systems with sources and receivers mounted permanently on the sea bottom is crucial. Akulichev et al. presented some experimental results from shallow zones of the World Ocean that served as a motive for developing a multifunction system with acoustic hardware and software. Here we present technical features and the sea test results of the system.

• Journal of Korea Multimedia Society
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• v.7 no.11
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• pp.1564-1570
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• 2004

Recently, The Analog TV has been being replaced by digital TV. This dues to the advantage of digital TV that sends data as well as audio/video stream to the receiver. But there are no standards or rules for processing the data. So, most companies or developers process the data in separate methods. This absence of standards or rules for processing the data causes many confusions and waste of time, money, man power. This Thesis proposes the solution of the problem not based on standardized respect but methodological respect. This solution is using XML in producing and playing the broadcasted multimedia contents and data. We believe that this solution saves much man power, time, money and increase reusability of expensive broadcasting contents.

• Journal of information and communication convergence engineering
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• v.13 no.4
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• pp.235-240
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• 2015

A decimation chain for multi-standard reconfigurable radios is presented for 900-MHz and 1,900-MHz dual-band cellular standards with a data interpolator based on the Lagrange method for adjusting the variable data rate to a fixed data rate appropriate for each standard. The two proposed configurations are analyzed and compared to provide insight into aliasing and the signal bandwidth by means of a newly introduced measure called interpolation error. The average interpolation error is reduced as the ratio of the sampling frequency to the signal BW is increased. The decimation chain and the multi-rate analog-to-digital converter are simulated to compute the interpolation error and the output signal-to-noise ratio. Further, a method to operate the above-mentioned chain under a compressed mode of operation is proposed in order to guarantee continuous packet connectivity for inter-radio-access technologies. The presented decimation chain can be applied to LTE, WCDMA, GSM multi-mode multi-band digital front-end which will ultimately lead to the software-defined radio.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2009.01a
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• pp.184-188
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• 2009

To interactively share High Definition (HD)-quality visualization over emerging ultra-high-speed network infrastructure, several lossless and low-delay real-time media (i.e., uncompressed HD video and audio) transport systems are being designed and prototyped. However, most of them still rely on expensive hardware components. As an effort to reduce the building cost of system, in this paper, we propose the integration of both transmitter and receiver machines into a single bi-directional transport system. After detailed bottleneck analysis and subsequent refinements of embedded software components, the proposed integration can provide Real-time Transport Protocol (RTP)-based bi-directional transport of uncompressed HD video and audio from a single machine. We also explain how to interface the Gbps-bandwidth display output of uncompressed HD media system to the networked tiled display of 10240 $\times$ 3200 super-high-resolution. Finally, to verify the feasibility of proposed integration, several prototype systems are built and evaluated by operating them in several different experiment scenarios.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.10 no.11
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• pp.5271-5289
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• 2016

On large-scale data analysis platforms deployed on cloud infrastructures over the Internet, the instability of the data transfer time and the dynamics of the processing rate require a more sophisticated data distribution scheme which maximizes parallel efficiency by achieving the balanced load among participated computing elements and by eliminating the idle time of each computing element. In particular, under the constraints that have the real-time and limited data buffer (in-memory storage) are given, it needs more controllable mechanism to prevent both the overflow and the underflow of the finite buffer. In this paper, we propose an auto regulated data provisioning model based on receiver-driven data pull model. On this model, we provide a synchronized data replenishment mechanism that implicitly avoids the data buffer overflow as well as explicitly regulates the data buffer underflow by adequately adjusting the buffer resilience. To estimate the optimal size of buffer resilience, we exploits an adaptive buffer resilience control scheme that minimizes both data buffer space and idle time of the processing elements based on directly measured sample path analysis. The simulation results show that the proposed scheme provides allowable approximation compared to the numerical results. Also, it is suitably efficient to apply for such a dynamic environment that cannot postulate the stochastic characteristic for the data transfer time, the data processing rate, or even an environment where the fluctuation of the both is presented.

• Journal of The Korean Astronomical Society
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• v.39 no.2
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• pp.51-56
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• 2006

We report the development of a semi-VLBI observation system operating at 21 cm and present the measurement of visibility function toward the sun using this system. The system consists of two 2.3 meter antennas with a maximum separation of 35 meter, a conventional high speed data acquisition system, and a set of programs for software correlation. Since two local oscillators of receiver modules are independent, data had to be fringe-fitted to yield the visibility amplitude. It is found that the visibility amplitude decreases and then bounces back as baseline increases. We confirm that solar disk with brighter limb best explains the measured visibility amplitude.