• Title/Summary/Keyword: high SNR

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Doppler-shift estimation of flat underwater channel using data-aided least-square approach

  • Pan, Weiqiang;Liu, Ping;Chen, Fangjiong;Ji, Fei;Feng, Jing
    • International Journal of Naval Architecture and Ocean Engineering
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    • v.7 no.2
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    • pp.426-434
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    • 2015
  • In this paper we proposed a dada-aided Doppler estimation method for underwater acoustic communication. The training sequence is non-dedicate, hence it can be designed for Doppler estimation as well as channel equalization. We assume the channel has been equalized and consider only flat-fading channel. First, based on the training symbols the theoretical received sequence is composed. Next the least square principle is applied to build the objective function, which minimizes the error between the composed and the actual received signal. Then an iterative approach is applied to solve the least square problem. The proposed approach involves an outer loop and inner loop, which resolve the channel gain and Doppler coefficient, respectively. The theoretical performance bound, i.e. the Cramer-Rao Lower Bound (CRLB) of estimation is also derived. Computer simulations results show that the proposed algorithm achieves the CRLB in medium to high SNR cases.

PARTICLE ACCELERATION IN SUPERNOVA REMNANTS

  • KANG, HYESUNG
    • Publications of The Korean Astronomical Society
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    • v.30 no.2
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    • pp.545-548
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    • 2015
  • Most high energy cosmic rays (CRs) are thought to be produced by diffusive shock acceleration (DSA) in supernova remnants (SNRs) within the Galaxy. Plasma and MHD simulations have shown that the self-excitation of MHD waves and amplification of magnetic fields via plasma instabilities are an integral part of DSA for strong collisionless shocks. In this study we explore how plasma processes such as plasma instabilities and wave-particle interactions can affect the energy spectra of CR protons and electrons, using time-dependent DSA simulations of SNR shocks. We demonstrate that the time-dependent evolution of the shock dynamics, the self-amplified magnetic fields and $Alfv{\acute{e}nic$ drift govern the highest energy end of the CR energy spectra. As a result, the spectral cutoffs in nonthermal X-ray and ${\gamma}$-ray radiation spectra are regulated by the evolution of the highest energy particles, which are injected at the early phase of SNRs. We also find that the maximum energy of CR protons can be boosted significantly only if the scale height of the magnetic field precursor is long enough to contain the diffusion lengths of the particles of interests. Thus, detailed understandings of nonlinear wave-particle interactions and time-dependent DSA simulations are crucial for understanding the nonthermal radiation from CR acceleration sources.

Study on DC-Offset Cancellation in a Direct Conversion Receiver

  • Park, Hong-Won
    • The Bulletin of The Korean Astronomical Society
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    • v.37 no.2
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    • pp.157.2-157.2
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    • 2012
  • Direct-conversion receivers often suffer from a DC-offset that is a by-product of the direct conversion process to baseband. In general, a basic approach to reduce the DC-offset is to do simple average of the baseband signal and remove the DC by subtracting the average. However, this gives rise to a residual DC offset which degrades the performance when the receiver adopts the coding schemes with high coding rates such as 8-PSK. Therefore, more advanced methods should be additionally required for better performance. While the training sequences are basically designed to have good auto-correlation properties to facilitate the channel estimation, they may be not good for the simultaneous estimation of the channel response and the DC-offset. Also the DC offset compensation under a bad condition does not give good results due to the estimation error. Correspondingly, the proposed scheme employs the two important points. First, the training sequence codes are divided into two groups by MSE(Mean Squared Errors) for estimating the channel taps and then SNR calculated from each group is compared to predefined threshold to do fine DC-offset estimation. Next, ON/OFF module is applied for preventing performance degradation by large estimation error under severe channel conditions. The simulation results of the proposed scheme shows good performances compared to the existing algorithm. As a result, this scheme is surely applicable to the receiver design in many communications systems.

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Contents Adaptive MCTF Using JND (JND를 이용한 적응적 MCTF)

  • Heo, Jae-Seong;Ryu, Chul
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.34 no.1C
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    • pp.48-55
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    • 2009
  • In scalable video coding, MCTF plays an important role for time-scalability and SNR-scalability. But there is image quality decreasing as MCTF level is increased because time interval of each frame is extended so that is hard to find suitable motion vector. In this paper, we propose an algorithm to prevent image quality from decreasing with unsuitable motion vector during MCTF update process using JND. We adapt JND to find errors within blocks of image and set a threshold which is used to add high frequency components during update process. We can overcome time-gap between frames and achieve better image quality through the proposed algorithm.

Implementation of High Speed LDPC Decode for Multi-Giga bps Cable Communication Service (Multi-Giga bps 케이블 통신 서비스를 위한 고속 LDPC 복호기 구현)

  • Jung, Joon-Young;Choi, Dong-Joon;Hur, Namho
    • Proceedings of the Korean Society of Broadcast Engineers Conference
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    • 2015.07a
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    • pp.301-302
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    • 2015
  • 케이블 방송망에서 멀티-Gbps(Giga bit per second) 초고속 인터넷 서비스 제공을 위해 최근 북미에서 DOCSIS 3.1(Data over Cable Service Interface Specifications Version 3.1) 표준을 발표하였다. DOCSIS 3.1 은 최대 10Gbps 하향 데이터 전송과 최대 2Gbps 의 상향 데이터 전송을 목표로 한다. DOCSIS 3.1 이 이전 DOCSIS 표준들과 다른 점은 전송 효율을 높이기 위해 물리계층 전송 방식에 큰 변화를 주었다는 점이다. 기존 6MHz 대역폭의 단일 반송파 전송 방식에서 최대 192MHz 광역 채널의 다중 반송파 전송 방식으로 변경하였다. 또한 채널 오류정정 방식으로 BCH(Bose, Chaudhuri, and Hocquenghem)와 LDPC(Low Density Parity Check) 연접부호를 적용하여, 이로 인한 SNR 성능 이득 통해 4096-QAM 의 고차 변조를 지원한다. 본 논문에서는 최대 192MHz 의 광역 채널로 전송되는 약 2Gbps 의 전송 데이터에 대한 채널 오류 정정을 위해 고속의 LDPC 복호기 구현 방법을 제시한다.

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An implementation of automated ECG interpretation algorithm and system(IV) - diagnosis parameter extractor and classifier (심전도 자동 진단 알고리즘 및 장치 구현(V) - 진단 파라미터 추출 및 진단기)

  • Kweon, H.J.;Jeong, K.S.;Lee, J.W.;Shin, K.S.;Lee, M.H.
    • Proceedings of the KOSOMBE Conference
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    • v.1996 no.05
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    • pp.298-302
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    • 1996
  • The representative beat with high SNR could be obtained by the signal averaging, correct and fast detection of significant points and waveform boundary could be obtained by adoption of search interval. All experimental results of waveform boundary were compared with CSE database which had the 5 referees results and 11 ECG measurement programs. All results were within tolerance made by referees, especially the end point of T wave were more close to the referee's results than other 11 measurement programs. The diagnosis parameters that might be used in the Minnsota code criteria were extracted from the representative beat. The diagnostic classification were fulfilled using Minnsota code criteria. Through the comparison on the diagnosis results from designed automated ECG analyzer(YECGA) and the results ECG analyzer manufactured by Fukuda denshi(FCG-2201) in Japan, reliance of the performance on designed system(YECGA) could be validated.

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A Scalable Audio Coder for High-quality Speech and Audio Services

  • Lee, Gil-Ho;Lee, Young-Han;Kim, Hong-Kook;Kim, Do-Young;Lee, Mi-Suk
    • MALSORI
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    • no.61
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    • pp.75-86
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    • 2007
  • In this paper, we propose a scalable audio coder, which has a variable bandwidth from the narrowband speech bandwidth to the audio bandwidth and also has a bit-rate from 8 to 320 kbits/s, in order to cope with the quality of service(QoS) according to the network load. First of all, the proposed scalable coder splits bandwidth of the input audio into narrowband up to around 4 kHz and above. Next, the narrowband signals are compressed by a speech coding method compatible to an existing standard speech coder such as G.729, and the other signals whose bandwidth is above the narrowband are compressed on the basis of a psychoacoustic model. It is shown from the objective quality tests using the signal-to-noise ratio(SNR) and the perceptual evaluation of audio quality(PEAQ) that the proposed scalable audio coder provides a comparable quality to the MPEG-1 Layer III (MP3) audio coder.

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Visualization of Surface Deformation on an Open-Hole Specimen Based on Grating Shearography

  • Lee, Jung-Ryul;Lee, Seung-Seok;Chung, Won-Seok
    • Journal of the Korean Society for Nondestructive Testing
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    • v.27 no.6
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    • pp.563-575
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    • 2007
  • This contribution contained the classical work of an open-hole tensile plate to demonstrate the performance of grating shearography and to compare with the results obtained by other full-field measurement techniques, The isotropic plate with an open-hole has often appeared in the previous contributions introducing novel full-field method and system. Grating shearography directly provided six quantitative measurands about the specimen's surface kinematics by using a single measurement set: three in-plane strains, in plane rotation, and two out-of-plane slopes. The quasi-plane wavefront of grating metrology led to high signal-to-noise ratio (SNR) and thus neither fitting nor filtering was applied, and the small shearing distance of $101{\mu}m$ could be used. The small shearing distance provided the outstanding spatial resolution of $80{\mu}m$ and sensitivity appropriate for experimental mechanics. Finally, the grating shearography enabled the visualization of the complex surface deformation around the hole and also detected parasitic flexions of the specimen in the micrometer regime during the tensile test.

Performance Analysis And Optimization For AF Two-Way Relaying With Relay Selection Over Mixed Rician And Rayleigh Fading

  • Fan, Zhangjun;Guo, Daoxing;Zhang, Bangning;Zeng, Li
    • KSII Transactions on Internet and Information Systems (TIIS)
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    • v.6 no.12
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    • pp.3275-3295
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    • 2012
  • In this paper, we analyze the performance of an amplify-and-forward (AF) two-way relaying system, where two sources exchange information via the aid of an intermediate relay that is selected among multiple relays according to max-min criterion. We consider a practical scenario, where one source-relay link undergoes Rician fading, and the other source-relay link is subject to Rayleigh fading. To be specific, we derive a tight lower bound for the outage probability. From this lower bound, the asymptotic outage probability and average symbol error rate (SER) expressions are derived to gain insight into the system performance at high signal-to-noise ratio (SNR) region. Furthermore, we investigate the optimal power allocation (PA) with fixed relay location (RL), optimal RL with fixed PA and joint optimization of PA and RL to minimize the outage probability and average SER. The analytical expressions are verified through Monte Carlo simulations, where the positive impact of Rician factor on the system performance is also illustrated. Simulation results also validate the effectiveness of the proposed PA and relay positioning schemes.

Implementation of Chip and Algorithm of a Speech Enhancement for an Automatic Speech Recognition Applied to Telematics Device (텔레메틱스 단말용 음성 인식을 위한 음성향상 알고리듬 및 칩 구현)

  • Kim, Hyoung-Gook
    • The Journal of The Korea Institute of Intelligent Transport Systems
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    • v.7 no.5
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    • pp.90-96
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    • 2008
  • This paper presents an algorithm of a single chip acoustic speech enhancement for telematics device. The algorithm consists of two stages, i.e. noise reduction and echo cancellation. An adaptive filter based on cross spectral estimation is used to cancel echo. The external background noise is eliminated and the clear speech is estimated by using MMSE log-spectral magnitude estimation. To be suitable for use in consumer electronics, we also design a low cost, high speed and flexible hardware architecture. The performance of the proposed speech enhancement algorithms were measured both by the signal-to-noise ratio(SNR) and recognition accuracy of an automatic speech recognition(ASR) and yields better results compared with the conventional methods.

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