• Title, Summary, Keyword: Signal to Noise Ratio

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Cramer-Rao Lower Bound of Effective Carrier-to-noise Power Ratio Estimation for a GPS L1 C/A Signal under Band-limited White Noise Jamming Environments (대역제한 백색잡음 재밍환경에서 GPS L1 C/A 신호를 위한 유효 반송파 대 잡음 전력비 추정치의 CRLB)

  • Yoo, Seungsoo;Yeom, Dong-Jin;Jee, Gyu-In;Kim, Sun Yong
    • Journal of Institute of Control, Robotics and Systems
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    • v.20 no.8
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    • pp.890-894
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    • 2014
  • In this paper, we derive the CRLB (Cramer-Rao Lower Bound) of effective carrier-to-noise power ratio ($C/N_0$) estimation for a GPS (Global Positioning System) L1 C/A (Coarse/Acquisition) signal under band-limited white noise jamming environments. The quality of a received GPS signal is commonly described in terms of its $C/N_0$, implying that the noise is white and thus can be described by scalar noise density. However, if some intentional interference is received to a victim GPS receiver, then the $C/N_0$ is no longer the efficacious performance indicator. The correct and straightforward measurement to analyze the receiving situation is the effective $C/N_0$. In this paper, we consider a band-limited white noise jamming whose bandwidth is 2MHz and is the same as one of the first null-to-null bandwidth of the GPS L1 C/A signal.

Effect of the Signal-to-Noise Power Spectra Ratio On MTF compensated EOC images

  • Kang, Chi-Ho;Choi, Hae-Jin
    • Proceedings of the KSRS Conference
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    • pp.202-207
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    • 2002
  • EOC (Electro-Optical Camera) of KOMPSAT-1 (Korea Multi-Purpose SATellite) has been producing land imageries of the world since January 2000. After image data are acquired by EOC, they are transmitted from satellite to ground via X-band RF signal. Then, EOC image data are generated and pass through radiometric and geometric corrections to generate standard products of EOC images. After radiometric correction on EOC image data, Modulation Transfer Function (MTF) compensation is applicable on EOC images with user's request for better image quality. MTF compensation is concerned with filtering EOC images to minimize the effect of degradations. For Image Receiving and Processing System (IRPE) at KOMPSAT Ground Station (KGS), Wiener filter is used in MTF compensation for EOC images. If the Pointing Spread Function (PSF) of EOC system is known, signal-to-noise power spectra ratio is the only factor in the determination of Wiener filter. In this paper, MTF compensation in IRPE at KGS is introduced and MTF compensated EOC 1R images are generated using Wiener filters with various signal-to-noise power spectra ratios. MTF compensated EOC 1R images are correlated with EOC 1R images for observing linearities between them. As a result, the effect of signal-to-noise power spectra ratio is shown on MTF compensated EOC 1R images.

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Recuction of the Influence of Background Noise in Sound Insulation Measurement (차음성능 측정에 있어서의 암소음의 영향의 저감 (1))

  • 염성곤;다치바나히데끼
    • Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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    • pp.495-498
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    • 2004
  • In the sound insulation measurements, the influence of background (extraneous) noise is often serious problem and how to reduce its effect and to improve the signal-to-noise(S/N) ratio is an important theme. As the background noise, such extraneous noises as road traffic noise and machine noise often disturb the measurement. In laboratory measurements on specimens with high sound insulation performances, even the internal noise of the measurement system can become a problem. To improve the signal-to-noise ratio and to improve the measurement accuracy, various kinds of digital signal processing techniques can be applied. In this paper, four kinds of digital signal processing techniques are applied and their effectiveness is examined by a simple sound insulation measurement.

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Recuction of the Influence of Background Noise in Sound Insulation Measurement (차음성능 측정에 있어서의 암소음의 영향의 저감 (2))

  • Yum, Sung-Gon;Tachibana, Hideki
    • Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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    • pp.441-444
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    • 2004
  • In the sound insulation measurements, the influence of background (extraneous) noise is often serious problem and how to reduce its effect and to improve the signal-to-noise(S/N) ratio is an important theme. As the background noise, such extraneous noises as road traffic noise and machine noise often disturb the measurement. In laboratory measurements on specimens with high sound insulation performances, even the internal noise of the measurement system can become a problem. To improve the signal-to-noise ratio and to improve the measurement accuracy, various kinds of digital signal processing techniques can be applied. In this paper, four kinds of digital signal processing techniques are applied and their effectiveness is examined through field measurements.

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Performance Enhancement of Speech Intelligibility in Communication System Using Combined Beamforming (directional microphone) and Speech Filtering Method (방향성 마이크로폰과 음성 필터링을 이용한 통신 시스템의 음성 인지도 향상)

  • Shin, Min-Cheol;Wang, Se-Myung
    • Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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    • pp.334-337
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    • 2005
  • The speech intelligibility is one of the most important factors in communication system. The speech intelligibility is related with speech to noise ratio. To enhance the speech to noise ratio, background noise reduction techniques are being developed. As a part of solution to noise reduction, this paper introduces directional microphone using beamforming method and speech filtering method. The directional microphone narrows the spatial range of processing signal into the direction of the target speech signal. The noise signal located in the same direction with speech still remains in the processing signal. To sort this mixed signal into speech and noise, as a following step, a speech-filtering method is applied to pick up only the speech signal from the processed signal. The speech filtering method is based on the characteristics of speech signal itself. The combined directional microphone and speech filtering method gives enhanced performance to speech intelligibility in communication system.

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Design of Deep De-nosing Network for Power Line Artifact in Electrocardiogram (심전도 신호의 전력선 잡음 제거를 위한 Deep De-noising Network 설계)

  • Kwon, Oyun;Lee, JeeEun;Kwon, Jun Hwan;Lim, Seong Jun;Yoo, Sun Kook
    • Journal of Korea Multimedia Society
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    • v.23 no.3
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    • pp.402-411
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    • 2020
  • Power line noise in electrocardiogram signals makes it difficult to diagnose cardiovascular disease. ECG signals without power line noise are needed to increase the accuracy of diagnosis. In this paper, it is proposed DNN(Deep Neural Network) model to remove the power line noise in ECG. The proposed model is learned with noisy ECG, and clean ECG. Performance of the proposed model were performed in various environments(varying amplitude, frequency change, real-time amplitude change). The evaluation used signal-to-noise ratio and root mean square error (RMSE). The difference in evaluation metrics between the noisy ECG signals and the de-noising ECG signals can demonstrate effectiveness as the de-noising model. The proposed DNN model learning result was a decrease in RMSE 0.0224dB and a increase in signal-to-noise ratio 1.048dB. The results performed in various environments showed a decrease in RMSE 1.7672dB and a increase in signal-to-noise ratio 15.1879dB in amplitude changes, a decrease in RMSE 0.0823dB and a increase in signal-to-noise ratio 4.9287dB in frequency changes. Finally, in real-time amplitude changes, RMSE was decreased 0.3886dB and signal-to-noise ratio was increased 11.4536dB. Thus, it was shown that the proposed DNN model can de-noise power line noise in ECG.

Research on Noise Reduction Algorithm Based on Combination of LMS Filter and Spectral Subtraction

  • Cao, Danyang;Chen, Zhixin;Gao, Xue
    • Journal of Information Processing Systems
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    • v.15 no.4
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    • pp.748-764
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    • 2019
  • In order to deal with the filtering delay problem of least mean square adaptive filter noise reduction algorithm and music noise problem of spectral subtraction algorithm during the speech signal processing, we combine these two algorithms and propose one novel noise reduction method, showing a strong performance on par or even better than state of the art methods. We first use the least mean square algorithm to reduce the average intensity of noise, and then add spectral subtraction algorithm to reduce remaining noise again. Experiments prove that using the spectral subtraction again after the least mean square adaptive filter algorithm overcomes shortcomings which come from the former two algorithms. Also the novel method increases the signal-to-noise ratio of original speech data and improves the final noise reduction performance.

Active noise control with on-line adaptive algorithm in a duct system (덕트에서 온라인 적응 알고리듬을 이용한 능동소음제어)

  • Kim, Heung-Seob;Hong, Jin-Seok;Oh, Jae-Eung
    • Transactions of the Korean Society of Mechanical Engineers A
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    • v.21 no.8
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    • pp.1332-1338
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    • 1997
  • In the case of the transfer function for the secondary path is dependent on time, the on-line method which can model it is continuously must be applied to the active noise control technique. And the adaptive random noise technique among the on-line methods is effective in the narrow-band control. In this method, the signal to noise ratio between random noise for modeling and primary noise is low. Therefore, the estimations of transfer function will be prone to inaccuracies and the convergence time will be too long. Such imperfections will have an influence upon the performance of an active noise controller. In this study, t enhance the signal to noise ratio, the on-line method that is combined the conventional adaptive random noise technique and the adaptive line enhancer, is proposed. By using proposed on-line method, a rigorous system identification and control of primary noise have been implemented.

A Feedback and Noise Cancellation Algorithm of Hearing Aids Using Dual Microphones (이중 마이크를 사용한 보청기의 궤환 및 잡음제거 알고리즘)

  • Lee, Haeng-Woo
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.36 no.7C
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    • pp.413-420
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    • 2011
  • This paper proposes a new adaptive algorithm to cancel the acoustic feedback and noise signals in the binaural hearing aids. The convergence performances of the proposed algorithm are improved by updating coefficients of the feedback canceller after the speech signal is cancelled from the residual signal with dual microphones. The feedback canceller firstly cancels the feedback signal from the microphone signal, and then the noise canceller reduces the noise by the beamforming method. To assure that binaural hearing aids converge stably, the left-sided hearing aid only is converged firstly, next the right-sided hearing aid only is converged. To verify performances of the proposed algorithm, simulations were carried out for a speech. As the results of simulations, it was proved that we can advance 14.43dB SFR(Signal to Feedback Ratio) on the average for the feedback canceller, 10.19dB SNR(Signal to Noise Ratio) improvement on the average for the noise canceller, in case that this algorithm is used.

An Analysis of a Phase Locked AM signal Detection (위상고정회로를 사용한 AM신호 검파방식의 해석)

  • 문상재
    • Journal of the Korean Institute of Telematics and Electronics
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    • v.13 no.5
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    • pp.24-29
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    • 1976
  • In the phase locked AM signal detection, phase locked loop is used to extract a synchronous carrier from an input AM signal. Under the assumption that input noise is white Gaussian and free-running frequency of voltage controlled oscillator is the same that of an input carrier, operational behaviours of phase locked loop is analyzed and signal to noise ratio of the detection is derived quentitatively. The results show that the phase locked AM signal detection method offers a higher degree of noise mmunity than conventional AM signal detections.

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