• Journal of the Korea Academia-Industrial cooperation Society
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• v.7 no.2
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• pp.206-212
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• 2006

Many image and video compression algorithms work by splitting the input image into blocks and producing variable-length coded bits for each block data. If variable-length coded data are transmitted consecutively, then the resulting coder is highly sensitive to channel errors. Therefore, most image and video techniques for providing some protection to the stream against channel errors usually involve adding a controlled amount of redundancy back into the stream. Such redundancy might take the form of resynchronization markers, which enable the decoder to restart the decoding process from the known state, in the event of transmission errors. The Error Resilient Entropy Code (EREC) is a well known method which can regain synchronization without any redundant information to convert from variable-length code to fixed-length code. This paper proposes an enhancement to EREC, which greatly improves its transmission ability for the compressed image quality without any redundant bits in the event of errors. The simulation result shows that the both objective and subjective quality of transmitted image is enhanced compared with the existing EREC at the same BER(Bit Error Rate).

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.13 no.2
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• pp.962-977
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• 2019

The human voice is a convenient method of information transfer between different objects such as between men, men and machine, between machines. The development of information and communication technology, the voice has been able to transfer farther than before. The way to communicate, it is to convert the voice to another form, transmit it, and then reconvert it back to sound. In such a communication process, a vocoder is a method of converting and re-converting a voice and sound. The CELP (Code-Excited Linear Prediction) type vocoder, one of the voice codecs, is adapted as a standard codec since it provides high quality sound even though its transmission speed is relatively low. The EVRC (Enhanced Variable Rate CODEC) and QCELP (Qualcomm Code-Excited Linear Prediction), variable bit rate vocoders, are used for mobile phones in 3G environment. For the real-time implementation of a vocoder, the reduction of sound quality is a typical problem. To improve the sound quality, that is important to know the size and shape of noise. In the existing sound quality improvement method, the voice activated is detected or used, or statistical methods are used by the large mount of data. However, there is a disadvantage in that no noise can be detected, when there is a continuous signal or when a change in noise is large.This paper focused on finding a better way to decrease the reduction of sound quality in lower bit transmission environments. Based on simulation results, this study proposed a preprocessor application that estimates the SNR (Signal to Noise Ratio) using the spectral SNR estimation method. The SNR estimation method adopted the IMBE (Improved Multi-Band Excitation) instead of using the SNR, which is a continuous speech signal. Finally, this application improves the quality of the vocoder by enhancing sound quality adaptively.

• The KIPS Transactions:PartC
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• v.15C no.5
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• pp.367-374
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• 2008

To realize the voice privacy between CDMA mobile phone and PSTN terminal, the voice frames shall be transmitted transparently between the heterogeneous networks. For satisfying this requirement, we propose the method which transmits voice frames using the CDMA circuit data channel in real time. In this paper we analyze the causes of voice delay which occurs during voice transmission using circuit data channel. And in order to overcome this kind of delay, the technique controlling the TCP control flag and the variable audio block construction algorithm according to the vocoder output rate are proposed. As a result of experimenting by applying the proposed method, we confirmed that the transit delay was improved with about average 70%.

• Journal of Sensor Science and Technology
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• v.27 no.1
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• pp.6-12
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• 2018

In this paper, a 32-variable pulse position modulation (32-VPPM) scheme is proposed to support a red-green-blue light-emitting-diode (RGB-LED)-based optical camera communication (OCC) system. Our proposed modulation scheme is designed to enhance the OCC data transmission rate, which is targeted for the wearable biomedical data monitoring system. The OCC technology has been utilized as an alternative solution to the radio frequency (RF) wireless system for long-term self-healthcare monitoring. Different biomedical signals, such as electrocardiograms, photoplethysmograms, and respiration signals are being monitored and transmitted wirelessly from the wearable biomedical device to the smartphone receiver. A common 30 frames per second (fps) smartphone camera with a CMOS image sensor is used to record a transmitted optical signal. Moreover, the overall proposed system architecture, modulation scheme, and data demodulation are discussed in this paper. The experimental result shows that the proposed system is able to achieve > 9 kbps using only a common smartphone camera receiver.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.7 no.5
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• pp.887-892
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• 2003

In this paper, we propose a simple scaling algorithm for CDMA mobile communications where a voice traffic signals are transmitted by individual one of several data rates at every frames. The traditional method is based on using look-up table called SMT(symbol metric table), but the proposed algorithm is real-time direct scaling method through simple bit manipulations without lookup table. The bit error rate performance is calculated by computer simulation over AWGN and Rayleigh fading channels. From the results, it is shown that the proposed algorithm outperforms the traditional SMT method on Rayleigh channel by 0.3∼0.8dB, while achieving the less H/W complexity.

• Journal of the Korea Society of Computer and Information
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• v.7 no.4
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• pp.127-132
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• 2002

This paper aims to adjust the window size according to the network condition that the sender determines by making the receiver participating in the congestion levels. TCP-RLDM has the measurement-based transmission strategy based on the data-receiving rate complementing TCP with the property of Additive Increase / Multiplicative Decrease. The protocol can make an performance improvement by responding differently according to the property of errors-whether congestion losses or transient transmission errors - to confront dynamically in heterogeneous environments with wired or wireless networks and delay-sensitive or -tolerant applications. By collecting data-receiving rate and the cause of errors from the receiver and by enabling sender to use the congestion avoidance strategy before occuring congestion possibly, the protocol works well at variable network environments.

• Proceedings of the KIEE Conference
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• 2002.11c
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• pp.210-213
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• 2002

This paper addresses the design and implementation of compensation of rain attenuation for Ka-band satellite communication system complied with DVB-RCS[2]. A structure of rain fade compensation scheme in the Ka-band satellite communication system is presented. Rain fade compensation scheme in this paper is mainly applied into return-link, which is the path through which user terminal transmit the data tn hub system providing a service. Symbol rate and code rate of channel code are used as variable transmission parameter for rain fade compensation. For estimation of channel environment, SNR of the user terminal which is measured by demodulator of hub system is used. Rain fade compensation scheme in the paper changes the symbol rate and/or code rate according to the measured SNR so that it can compensate the attenuation of the signal.

• Journal of Communications and Networks
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• v.4 no.2
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• pp.108-117
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• 2002

TCP, which was developed on the basis of wired links, supposes that packet losses are caused by network congestion. In a wireless network, however, packet losses due to data corruption occur frequently. Since TCP does not distinguish loss types, it applies its congestion control mechanism to non-congestion losses as well as congestion losses. As a result, the throughput of TCP is degraded. To solve this problem of TCP over wireless links, previous researches, such as split-connection and end-to-end schemes, tried to distinguish the loss types and applied the congestion control to only congestion losses; yet they do nothing for non-congestion losses. We propose a novel transport protocol for wireless networks. The protocol called VS-TCP (Variable Segment size Transmission Control Protocol) has a reaction mechanism for a non-congestion loss. VS-TCP varies a segment size according to a non-congestion loss rate, and therefore enhances the performance. If packet losses due to data corruption occur frequently, VS-TCP decreases a segment size in order to reduce both the retransmission overhead and packet corruption probability. If packets are rarely lost, it increases the size so as to lower the header overhead. Via simulations, we compared VS-TCP and other schemes. Our results show that the segment-size variation mechanism of VS-TCP achieves a substantial performance enhancement.

• The KIPS Transactions:PartC
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• v.10C no.4
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• pp.479-484
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• 2003

In the current Internet, it is difficult for an Internet Phone to guarantee the QoS due to variable network conditions such as packet loss rate, delay and bandwidth. In addition, the QoS of an Internet Video Phone is more hard to guarantee because of video data. In this paper, we investigate application-level QoS control schemes that can adapt to variable network conditions, and describe an error control scheme and a congestion control scheme. Based on these QoS control schemes, we have designed and implemented an Internet Video Phone System that supports adaptive audio and video delivery. Through experiments, we found that the Internet Video Phone can reduce the packet loss rate considerably as well as adjust the transmission rate considering other TCP flows.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.10 no.2
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• pp.159-165
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• 2010

The Multi-code (Mc) modulation has been developed for high-speed data transmission over the wireless environments, but it suffers two critical problems due to the limited resource of Orthogonal Codes (OC) and high Peak-Average Power Ratio (PAPR). In this paper, we propose a novel modulation technique called AOCG [1] (Advanced Orthogonal Code Group)-OFDM (Orthogonal Frequency Division Multiplexing) to solve the above problems and obtain the variable high bit rates which can be controlled by the four parameters depending on the quality of services (QoS) required by users.