• KSII Transactions on Internet and Information Systems (TIIS)
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• v.8 no.9
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• pp.3110-3125
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• 2014

Network coding improves TCP's performance in lossy wireless networks. However, the complex congestion window evolution of network coded TCP (TCP-NC) makes the analysis of end-to-end throughput challenging. This paper analyzes the evolutionary process of TCP-NC against lossy links. An analytic model is established by applying a two-dimensional Markov chain. With maximum window size, end-to-end erasure rate and redundancy parameter as input parameters, the analytic model can reflect window evolution and calculate end-to-end throughput of TCP-NC precisely. The key point of our model is that by the novel definition of the states of Markov chain, both the number of related states and the computation complexity are substantially reduced. Our work helps to understand the factors that affect TCP-NC's performance and lay the foundation of its optimization. Extensive simulations on NS2 show that the analytic model features fairly high accuracy.

• Journal of Korea Multimedia Society
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• v.3 no.2
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• pp.164-173
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• 2000

This paper presents an efficient bandwidth allocation method for multiple source in the two-layer video coding of different spatial resolution. We first investigate the model of bitrate distortion in the MPEG-2 spacial scalable coding,. By using approximated model parameters, than we propose an efficient bitrate control method in order to keep the same distortion level among coders and the constant quality ratio between layers. Simulation results show that the proposed method can satify the user requirement in comparison to the conventional method.

• Journal of the Korean Institute of Telematics and Electronics
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• v.23 no.4
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• pp.429-438
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• 1986

In this paper we have designed and implemented a real-time, full-duplex SBC (sub-band coding) codec at 16kbps using a high speed digital signal processor, NEC 7720. The SBC codec employs a QMF(quadrature mirror filter) filter bank based on the tree structures of two-band analysis-synthesis pairs to partition speech signal into 4 octabe bands. Computer simulation has been done to investigate the effect of fixed-point computation of the NEC 7720. Three different performance measures, the conventional signal-to-noise ratio, the informal listening test, and an LPC(linear predictive coding)distance measure, have been used in this simulation. The necessary parameters have been optimized through the simulation. The developed hardware and software have been tested in real-time operation using a hardware emulator.

• Proceedings of the IEEK Conference
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• 1999.06a
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• pp.1107-1110
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• 1999

In speech signal processing, the accurate decision of the voiced/unvoiced sound is important for robust word recognition and analysis and a high coding efficiency. In this paper, we propose the mehod of the voiced/unvoiced decision using the LSP parameter which represents the spectrum characteristics of the speech signal. The voiced sound has many more LSP parameters in low frequency region. To the contrary, the unvoiced sound has many more LSP parameters in high frequency region. That is, the LSP parameter distribution of the voiced sound is different to that of the unvoiced sound. Also, the voiced sound has the minimun value of sequantial intervals of the LSP parameters in low frequency region. The unvoiced sound has it in high frequency region. we decide the voiced/unvoiced sound by using this charateristics. We used the proposed method to some continuous speech and then achieved good performance.

• Proceedings of the KIEE Conference
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• 2004.05a
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• pp.68-70
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• 2004

In this paper we develop an adaptive interface between video compression and transission protocols to handle QoS fluctuations that are common to mobile communication systems. We consider various generic design alternatives for QoS adaptation and identify 'QoS negotiation' as the most promissing. This method gives the best possibilities to obtain system-wide efficiency. To handle the indued system complexity we apply a design philosophy (called ARC) that separates implementation dependencies by introducing QoS interfaces between system modules. In the ARC phlosophy the implementation details are hidden in the subsystems. To assure efficient adaptation, the QoS must be negotiated between modules. We select the QoS parameters that are both necessary and sufficient for efficient negotiation between the video encoder and protocol modules. We describe the relation between the QoS parameters at the interface and the internal parameters of common video coding methods and protocol elements. Furthermore, we describe a negotiation procedure that allows a system-wide optimum to emerge.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.32 no.9C
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• pp.840-848
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• 2007

A multiple description coding (MDC) scheme for 3-D Data is presented. First, a plane-based 3-D data is split into two descriptions, each of which has identical contribution in 3-D surface reconstruction. In order to maximize the visual quality of reconstructed 3-D data, then, plane parameters are modified according to channel error condition. Finally, these descriptions are compressed and transmitted over distinct channels. In decoder, if two descriptions are available, we reconstruct a high quality 3-D data. If only one description is transmitted, however, 3-D surface recovery scheme reduces artifacts on erroneous 3-D surface, yielding a smooth 3-D surface. Therefore, the proposed algorithm guarantees acceptable quality reconstruction of 3-D data even though one channel is totally lost.

• Journal of Communications and Networks
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• v.2 no.1
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• pp.76-84
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• 2000

Communication systems, including cell-based mobile communication systems, multiple satellite communication systems of multi-beam satellite systems, require reliable handoff methods between cell-to-cell, satellite-to-satellite of beam-to-team, respectively. Recent measurement of a CDMA cellular system indicates that the system is in handoff at about 35% to 70% of an average call period. Therefore, system reliability during handoff is one of the major system performance parameters and eventually becomes a factor in the overall system capacity. This paper presents novel and improved techniques for handoff in cellular communications, multi-beam and multi-satellite systems that require handoff during a session. this new handoff system combines the soft handoff mechanism currently implemented in the IS-95 CDMA with code and packet diversity combining techniques and an iterative decoding algorithm (Turbo Coding). the Turbo code introduced by Berrou et all. has been demonstrated its remarkable performance achieving the near Shannon channel capacity [1]. Recently. Turbo codes have been adapted as the coding scheme for the data transmission of the third generation international cellular communication standards : UTRA and CDMA 2000. Our proposed encoder and decoder schemes modified from the original Turbo code is suitable for the code and packet diversity combining techniques. this proposed system provides not only an unprecedented coding gain from the Turbo code and it iterative decoding, but also gain induced by the code and packet diversity combining technique which is similar to the hybrid Type II ARQ. We demonstrate performance improvements in AWGN channel and Rayleigh fading channel with perfect channel state information (CSI) through simulations for at low signal to noise ratio and analysis using exact upper bounding techniques for medium to high signal to noise ratio.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.47 no.2
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• pp.11-19
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• 2010

Most of deblocking filters assumes that the block boundaries are accurately known and the coding information like quantization parameters are available. In some applications such as commercial television, however, many external video inputs without known block boundary and coding information arc given. If a decompressed video sequence heavily degraded with blocking artifacts is given through the external video port, it is absolutely necessary to detect block boundaries and control the strength of deblocking filtering by analysing the given images. This paper presents an efficient method to find the block boundaries and estimate the strength of the blocking artifacts without the knowledge of coding information. In addition, the confidence of the estimated blocking artifact information is also evaluated to control the adaptive deblocking filter robustly. Experiment results show that the estimated block boundary locations and strength relative strength and confidence information are practically good enough to reduce the blocking artifacts without prior knowledge.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.3C
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• pp.248-256
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• 2003

Space-time coding was designed for an efficient transmit diversity technique to improve performance of wireless communication. For the transmit diversity using space-time coding, the receiver requires to estimate channel parameters corresponding to each transmit antennas. In this paper, we propose an efficient channel estimation scheme based on trigonometric polynomial approximation in OFDM systems with transmit diversity using space-time coding. The proposed scheme is more efficient than the conventional scheme in terms of the computational complexity. For QAM modulation, when the size of FFH is 128, the conventional scheme with significant tap caching of 7 requires 9852 complex multiplications for TU, HT and BU channels. But the proposed scheme requires 2560, 7680 and 3584 complex multiplications for TU, HT and BU channels, respectively. Especially, for channels with smaller Doppler frequency and delay spreads, the proposed scheme has the improved BER performance and complexity. In addition, we evaluate the performance of maximum delay spread estimation in unknown channel. The performance of the proposed scheme is investigated by computer simulation in various multi-path fading environments.

• Proceedings of the Acoustical Society of Korea Conference
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• 1994.06a
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• pp.893-898
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• 1994

In this paper, the influence of voice source estimation and modeling on speech synthesis and coding is examined and then their new estimation and modeling techniques are proposed and verified by computer simulation. It is known that the existing speech synthesizer produced the speech which is dull and inanimated. These problems are arised from the fact that existing estimation and modeling techniques can not give more accurate voice parameters. Therefore, in this paper we propose a new voice source estimation algorithm and modeling techniques which can not give more accurate voice parameters. Therefore, in this paper we propose a new voice source estimation algorithm and modeling techniques which can represent a variety of source characteristics. First, we divide speech samples in one pitch region into four parts having different characteristics. Second, the vocal-tract parameters and voice source waveforms are estimated in each regions differently using sequential SVD. Third, we propose composite source model as a new voice source model which is represented by weighted sum of pre-defined basis functions. And finally, the weights and time-shift parameters of the proposed composite source model are estimeted uning EM(estimate maximize) algorithm. Experimental results indicate that the proposed estimation and modeling methods can estimate more accurate voice source waveforms and represent various source characteristics.