• Journal of the Institute of Convergence Signal Processing
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• v.5 no.3
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• pp.216-222
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• 2004

Frequency selective fading is a limiting factor for transmission rate and performance in high-speed multimedia communications. In this paper, we propose a turbo coded adaptive quadrature amplitude modulation (QAM) system for efficient high-speed transmission. By making use of a user-friendly simulation platform of SPW, the proposed turbo coded adaptive QAM system(TuAQAM) is developed and its performance is evaluated in terms of throughput and BER performance. Two channel models having delay spreads of 700ns and 1400ns are created for the simulations. It is shown that the proposed TuAQAM system provides a performance improvement of approximately 3dB and improved throughput for the channel model whose delay spread is 700ns. Similarly, a performance improvement is also achieved for the channel model whose delay spread is 1400ns.

• Journal of the Institute of Electronics and Information Engineers
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• v.54 no.8
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• pp.26-33
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• 2017

Vehicle communication can facilitate efficient coordination among vehicles on the road and enable future vehicular applications such as vehicle safety enhancement, infotainment, or even autonomous driving. In the $3^{rd}$ Generation Partnership Project (3GPP), many studies focus on long term evolution (LTE)-based vehicle communication. Because vehicle speed is high enough to cause severe channel distortion in vehicle-to-vehicle (V2V) environments. We can utilize channel estimation methods to approach a reliable vehicle communication systems. Conventional channel estimation schemes can be categorized as least-squares (LS), decision-directed channel estimation (DDCE), spectral temporal averaging (STA), and smoothing methods. In this study, we propose a smart channel estimation scheme in LTE-based V2V environments. The channel estimation scheme, based on an LTE uplink system, uses a demodulation reference signal (DMRS) as the pilot symbol. Unlike conventional channel estimation schemes, we propose an adaptive smoothing channel estimation scheme (ASCE) using quadratic smoothing (QS) of the pilot symbols, which estimates a channel with greater accuracy and adaptively estimates channels in data symbols. In simulation results, the proposed ASCE scheme shows improved overall performance in terms of the normalized mean square error (NMSE) and bit error rate (BER) relative to conventional schemes.

• Proceedings of the IEEK Conference
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• 2000.06d
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• pp.92-97
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• 2000

An AMR(Adaptive Multi-Rate) speech coding algorithm has been adopted as a standard speech codec for IMT-2000. It is based on the algebraic CELP, and consists of eight speech coding modes having the bit rate from 4.75 kbit/s to 12.2 kbit/s. It also contains the VAD(Voice Activity Detector), SCR (Source Controlled Rate) operation, and error concealment scheme for robustness in a radio channel. The bit rate of AMR is changed on a frame basis depending on the channel condition. In this paper, we introduced AMR speech coding algorithm and performed the real-time implementation using TMS320C6201, i.e., a Texas Instrument's fixed-point DSP. With the ANSI C source code released from ETSI and 3GPP, we convert and optimize the program to make it run in real time using the C compiler and assembly language. It is verified that the decoded result of the implemented speech codec on the DSP is identical with the PC simulation result using ANSI C code for test sequences. Also, actual sound input/output test using microphone and speaker demonstrates its proper real-time operation without distortions or delays.

• Journal of the Korean Institute of Telematics and Electronics B
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• v.33B no.6
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• pp.45-53
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• 1996

In this paper, the convergence behavior of timing phase when an adaptive partial response equalizer and decision-directed type of a PLL work together in a digital recording channel is described. The phenomena of getting biased in timing phase when the convergence parameter of an adaptive partial response equalizer and timing recovery constant of a PLL are not selected properly is introduced. The phenomena, occurring due to perturbation of timing phase, are analyzed, by computer simulation and the region of ocnvergence for timing phase is discussed. Also, a method to overcome the phenomena using a variable step-size parameter is described.

• Journal of Biomedical Engineering Research
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• v.22 no.3
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• pp.285-291
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• 2001

In this paper, we designed two preprocessing adaptive filter - high pass filter and notch filter - using FPGA. For minimizing the calculation load of multi-channel and high-resolution ECG system, we utilize FPGA rather than digital signal processing chip. To implement the designed filters in FPGA, we utilize FPGA design tool(Altera corporation, MAX-PLUS II) and CSE database as test data. In order to evaluate the performance in terms of processing time, we compared the designed filters with the digital filters implemented by ADSP21061(Analog Devices). As a result, the filters implemented by FPGA showed better performance than the filters based on ADSP21061. As a consequence of examination, we conclude that FPGA is a useful solution in multi-channel and high-resolution signal processing.

• Journal of Advanced Navigation Technology
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• v.11 no.1
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• pp.37-42
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• 2007

In this paper, we analyzed the performance of OFDM systems with adaptive equalizer that considers the frequency offset, the frequency non-selective fading, and two-path microwave Rummer's model channels. First of all, it is analyzed that the performance degradation, which is caused by the offset and the non-selective fading channel, through simulation. As the results of the simulation, the performance of the OFDM system is greatly influenced by the offset and channels. The more the frequency offset is, the worse the performance of the OFDM system is. However, if the adaptive equalizer is adopted to the OFDM system, the performance is enhanced up to the limited rang.

• International Journal of Internet, Broadcasting and Communication
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• v.12 no.2
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• pp.21-29
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• 2020

This paper presents an effective single image haze removal using edge-preserving and adaptive transmission estimation to enhance the visibility of outdoor images vulnerable to weather and environmental conditions with computational complexity reduction. The conventional methods involve the time-consuming refinement process. The proposed transmission estimation however does not require the refinement, since it preserves the edges effectively, which selects one between the pixel-based dark channel and the patch-based dark channel in the vicinity of edges. Moreover, we propose an adaptive transmission estimation to improve the visual quality particularly in bright areas like sky. Experimental results with various hazy images represent that the proposed method is superior to the conventional methods in both subjective visual quality and computational complexity. The proposed method can be adopted to compose a haze removal module for realtime devices such as mobile devices, digital cameras, autonomous vehicles, and so on as well as PCs that have enough processing resources.

• Journal of Communications and Networks
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• v.16 no.3
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• pp.322-334
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• 2014

Multimedia content is very sensitive to packet loss and therefore multimedia streams are typically protected against packet loss, either by supporting retransmission requests or by adding redundant forward error correction (FEC) data. However, the redundant FEC information introduces significant additional bandwidth requirements, as compared to the bitrate of the original video stream. Especially on wireless and mobile networks, bandwidth availability is limited and variable. In this article, an adaptive FEC (A-FEC) system is presented whereby the redundancy rate is dynamically adjusted to the packet loss, based on feedback messages from the client. We present a statistical model of our A-FEC system and validate the proposed system under different packet loss conditions and loss probabilities. The experimental results show that 57-95%bandwidth gain can be achieved compared with a static FEC approach.

• Journal of the Korean Institute of Telematics and Electronics A
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• v.31A no.12
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• pp.1-8
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• 1994

In this paper, we hve compare the performance of DAR(Dynamic Adaptive Routing) with that of FSR(Flooding Search Routing) to select an adequate routing protocol in circuit-switched networs. As a performance factor, we have considered call setup time, which is the key factor of performance evaluation in circuit switched networks. We have evaluated the performance of two methods in grid topology circuit-switched networks using a commn channel signaling scheme, as application examples. As results, FSR method shows better performance than DAR method under light traffic load, when the number of links by which call has passed increases, but DAR method represents better performance than FSR method under heavy traffic load or large networks because of redundant packets.

• Journal of the Korean Institute of Telematics and Electronics A
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• v.31A no.12
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• pp.9-15
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• 1994

In this paper, the linearized power amplifier using digital adaptive predistorter is implemented in order to restrict spectral spreading and adjacent channel interference. The linearized systems is composed of a DSP56001 processor that executes predistortion in baseband. 90.deg. phase shifter, power splitter/combiner, quadrature modulator/demodulator of 360MHz band, and nonlinear amplifier. A ${\pi}$/4-shift QPSK is used to modulate digital random signals. As the quantized power of baseband signal and the output of amplifier are fed to the predistorter, and predistorting values are calculated using an adaptive algorithm. In the experiment, a peak to sidelobe ratio of the linearized amplifier is improved up to 15dB in comparison with conventional nonlinear amplifier, which means that the distortion of transmitted signal is decreased and adjacent channel interference was reduced.