• Title/Summary/Keyword: SNR 추정

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A Study about Performance of Sum-Product Decoder Considering Adaptive Bit-Loading in LDPC Coded OFDM Systems (LDPC Coded OFDM 시스템에서 적응형 비트 로딩을 고려한 Sum-Product 복호기 성능에 관한 연구)

  • Oh, Hui-Myoung;Kim, Young-Sun;Lee, Jae-Jo
    • Proceedings of the KIEE Conference
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    • 2006.07d
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    • pp.2027-2028
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    • 2006
  • 추정된 채널 정보를 바탕으로 적용하는 적응형 비트 로딩 방식은, 전력선 통신 시스템의 고속화 및 대용량 데이터 전송을 위해 최근 대두되고 있는 LDPC(Low Density Parity Check) coded OFDM 시스템에 대해, 한정된 주파수 대역과 신호 전력의 효율적 사용을 제공한다. 그러나 적응형 비트로딩 방식은 한정된 수의 일정 SNR(신호대 잡음 전력비) 구간에 대한 mapping 방식으로 적용되기 때문에 송수신 과정에서 추정된 채널 정보를 이용하는 sum-product 복호기가 채널 변화에 민감하게 반응하지 못하는 상황이 발생하며, 결국 송신단에서 채널 추정 결과를 바탕으로 선택된 SNR 범위에 대해서는 실제 수신되는 신호에 대한 SNR과의 차이가 존재하고 시스템의 성능은 그 만큼의 성능 열하로 나타나게 된다. 본 논문에서는 이러한 성능 열하 정도를 시뮬레이션을 통해 확인하였다.

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Improved Timing Synchronization Using Phase Difference between Subcarriers in OFDMA Uplink Systems (OFDMA 상향 링크 시스템에서 부반송파간 위상 회전 정보를 이용한 개선된 시간 동기 추정 알고리즘)

  • Lee, Sung-Eun;Hong, Dae-Sik
    • Journal of the Institute of Electronics Engineers of Korea TC
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    • v.46 no.2
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    • pp.46-52
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    • 2009
  • In this paper, the timing estimator based on the principle of the best linear unbiased estimator (BLUE) is proposed in OFDMA uplink systems. The proposed timing estimator exploits the phase information of the differential correlation between adjacent subcarriers. The differential correlation can extract the information about timing offset and mitigate the distortion of the signal caused by the frequency selectivity of channel. Compared with conventional methods, the proposed estimator shows more accurate capability in estimation. In addition, the estimator is hardly affected by the distortion caused by the frequency selectivity of channel. Simulation results confirm that the proposed estimator shows a small error mean and a relatively small error variance. In addition, the performance of the estimator is evaluated by means of SNR loss. It is shown by simulations that the SNR loss of the proposed estimator by estimation errors is less than 0.4 dB for the SNR values between 0 and 20 dB. This might indicate that the proposed estimator is suitable for the timing synchronization of multiple users in OFDMA uplink systems.

A Study on Estimation of Beat Frequency in a Water Level Measurement Radar (수위 측정 레이다에서의 비트 주파수 추정에 관한 연구)

  • Lee, Jonggil
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.18 no.8
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    • pp.1791-1797
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    • 2014
  • A water level measurement radar estimates beat frequencies to extract the corresponding water level information for water resource management. Therefore, it is required to maintain the high degree of reliability and accuracy in beat frequency estimates. However, Beat spectra of water surface return echoes can have very widely varying shapes according to system frequency linearity, measurement environments and weather conditions. Therefore, serious problems may arise in maintaining the reliability and accuracy of the conventional level estimation method. Therefore, in this paper, high resolution spectrum estimates are analyzed for improvement of the conventional method. These methods show that the more accurate level measurement can be possible on the condition that SNR is higher than a certain required threshold. However, if SNR is too low, the conventional method shows that estimates are more reliable than the suggested method though absolute errors are too large.

A Carrier Frequency Offset Estimation Algorithm for IEEE802.11n system (IEEE802.11n 시스템에 적용가능한 반송파 주파수 옵셋 추정 알고리즘)

  • Jung, Hyeok-Koo
    • Journal of the Institute of Electronics Engineers of Korea TC
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    • v.45 no.5
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    • pp.21-29
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    • 2008
  • This paper proposes a carrier frequency of set estimation algorithm for IEEE802.11n system. As IEEE802.11n is a multiple input multiple output(MIMO) system, so there are several combining techniques which are used in multiple receive antenna system. In this paper, we propose hybrid carrier frequency offset estimation algorithms using combining techniques in multiple receive antenna systems, and show that the proposed selection combining carrier frequency offset (CFO) estimation algorithm can estimate carrier frequency offset within 1/10 MSE error at SNR 10 dB in channel B and within 1/2 MSE error at SNR 10 dB in channel D rather than the conventional MIMO CFO one.

Peformance Comparisons of CDMA, OFDM, and MC-CDMA with Inaccurate Channel Estimates and Low-SNR Environments (신호대잡음비가 낮고 부정확한 채널추정값을 가질 때의 CDMA, OFDM, MC-CDMA의 성능 비교)

  • Rim Minjoong
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.30 no.1A
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    • pp.55-61
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    • 2005
  • Mobile communication systems are required not only to support high-data-rate transmissions in favorable channel conditions but also to be able to tolerate hostile environments possibly encountered by cellular communications. This paper compares CDMA(Code Division Multiple Access), OFDM(Orthogonal Frequency Division Multiplexing), and MC-CDMA(Multi-Carrier CDMA)with inaccurate channel estimates and low SNR environments. The equations and simulation results show that the performance losses in CDMA systems due to imprecise channel estimates are not considerable while frequency-hopping of OFDM systems can result in more than l0dB SNR losses. Also, this paper show that frequency-spreading of MC-CDMA can be very helpful for channel compensation performances than frequency-hopping or time-spreading of OFDM.

Two-step a priori SNR Estimation in the Log-mel Domain Considering Phase Information (위상 정보를 고려한 로그멜 영역에서의 2단계 선험 SNR 추정)

  • Lee, Yun-Kyung;Kwon, Oh-Wook
    • Phonetics and Speech Sciences
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    • v.3 no.1
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    • pp.87-94
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    • 2011
  • The decision directed (DD) approach is widely used to determine a priori SNR from noisy speech signals. In conventional speech enhancement systems with a DD approach, a priori SNR is estimated by using only the magnitude components and consequently follows a posteriori SNR with one frame delay. We propose a phase-dependent two-step a priori SNR estimator based on the minimum mean square error (MMSE) in the log-mel spectral domain so that we can consider both magnitude and phase information, and it can overcome the performance degradation caused by one frame delay. From the experimental results, the proposed estimator is shown to improve the output SNR of enhanced speech signals by 2.3 dB compared to the conventional DD approach-based system.

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A Novel Approach to a Robust A Priori SNR Estimator in Speech Enhancement (음성 향상에서 강인한 새로운 선행 SNR 추정 기법에 관한 연구)

  • Park, Yun-Sik;Chang, Joon-Hyuk
    • The Journal of the Acoustical Society of Korea
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    • v.25 no.8
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    • pp.383-388
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    • 2006
  • This Paper presents a novel approach to single channel microphone speech enhancement in noisy environments. Widely used noise reduction techniques based on the spectral subtraction are generally expressed as a spectral gam depending on the signal-to-noise ratio (SNR). The well-known decision-directed(DD) estimator of Ephraim and Malah efficiently reduces musical noise under the background noise conditions, but generates the delay of the a prioiri SNR because the DD weights the speech spectrum component of the Previous frame in the speech signal. Therefore, the noise suppression gain which is affected by the delay of the a priori SNR, which is estimated by the DD matches the previous frame rather than the current one, so after noise suppression. this degrades the noise reduction performance during speech transient periods. We propose a computationally simple but effective speech enhancement technique based on the sigmoid type function for the weight Parameter of the DD. The proposed approach solves the delay problem about the main parameter, the a priori SNR of the DD while maintaining the benefits of the DD. Performances of the proposed enhancement algorithm are evaluated by ITU-T p.862 Perceptual Evaluation of Speech duality (PESQ). the Mean Opinion Score (MOS) and the speech spectrogram under various noise environments and yields better results compared with the fixed weight parameter of the DD.

Decision Feedback Doppler Adaptive Band-Limit Algorithm for Maximum Doppler frequency Estimation (속도 추정 시 부가 잡음의 영향을 억제하기 위한 결정 궤환 적응형 대역 제한 방법에 대한 연구)

  • 박구현;한상철;류탁기;홍대식;강창언
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.28 no.11C
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    • pp.1111-1117
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    • 2003
  • The maximum Doppler frequency, or equivalently, the mobile speed is very useful information to optimize the performance of many wireless communication systems. However, the performance of a maximum Doppler frequency estimator is limited since it requires an estimate of the signal-to-noise ratio (SNR) of the channel environment. In this paper, the improved method for the maximum Doppler frequency estimations based on the decision feedback Doppler adaptive band-limit (DF-DABL) method is proposed. To reduce the effect of additive noise, the proposed algorithm uses a novel Doppler adaptive band-limit (DABL) technique. The distortion due to the additive noise is drastically removed by the proposed DF-DABL method. Especially, the DF-DABL method does not need any other channel information such as SNR.

Adaptive Spectral Subtraction Method Using SNR and Masking Effect for Robust Speech Recognition in Noisy Environments (잡음환경에 강인한 음성인식을 위해 SNR과 마스킹 효과를 이용한 적응 스펙트럼 차감법)

  • 김태준;김종훈;이경모;이정현
    • Proceedings of the Korean Information Science Society Conference
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    • 2004.10b
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    • pp.580-582
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    • 2004
  • 스펙트럼 차감과정에서 발생하는 잔류 잡음을 제거하는 방법으로 파라메터를 이용하는 적응 스펙트럼 차감법이 있다. 이는 파라메터를 증가시켜 잔류 잡음을 감소시키는 방법이지만 파라메터를 과도하게 증가시킬 경우 음성 왜곡이 발생한다. 따라서, 적절한 파라메터를 추출하기 위하여 SNR이나, 마스킹 효과 등을 이용한 방법들이 제안되었으나 과도한 잡음의 제거로 인한 음성 왜곡 문제와 낮은 SNR에서 부정확한 파라메터의 추출 문제는 여전히 해결해야 할 과제로 남아있다. 본 논문은 기존의 SNR을 이용한 방법에 마스킹 효과를 적용한 수정된 적응 스펙트럼 차감법을 제안한다. 제안된 방법에서는 마스킹 임계치를 이용하여 잡음 추정값을 재 계산 항으로써 SNR을 향상시켰고, 이를 이용하여 파라메터를 추출함으로써 성능을 개선했다 성능평가 결과, 제안한 차감법을 적용한 음성신호를 고립단어 음성인식 시스템에 적용했을 때 기존의 방법 보다 인식률이 향상된 것을 확인할 수 있었다.

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Performance Improvement of Perceptual Filter Using Noise Energy Control (잡음 에너지 제어를 통한 지각 필터 성능 개선)

  • Seo Joung-Kook;Cha Hyung-Tai
    • The Journal of the Acoustical Society of Korea
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    • v.24 no.1
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    • pp.43-51
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    • 2005
  • In this paper, we propose an algorithm that improves a tone quality of a noisy audio signal in order to enhance a Performance of perceptual filter using noise energy control. Most of the algorithms which were proposed by the other researchers usually applied a filter using the noise energy acquired from a silent range. In this case. the improvement rate of tone quality decreases if the noise energy is changed by the magnitude or environment variation in a signal frame. But the Proposed method Provides the means to find a food estimated noise through energy control of the estimated noise which is obtained from a silent range. Also we can get the enhancement of tone qualify in low frequency band unlike other methods. To show the performance of the Proposed algorithm, various input signals which had a different signal-to-noise ratio (SNR) such as 5dB, l0dB, 15dB and 20dB were used to test the proposed algorithm. With the proposed algorithm, we could confirm the enhancement of tone quality in terms of segmental SNR (SSNR). noise-to-mask ration (NMR) and mean opinion score (MOS) test.