• Title/Summary/Keyword: Signal-to-noise Ratio

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A New Noise Reduction Method Based on Linear Prediction

  • Kawamura, Arata;Fujii, Kensaku;Itho, Yoshio;Fukui, Yutaka
    • Proceedings of the IEEK Conference
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    • 2000.07a
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    • pp.260-263
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    • 2000
  • A technique that uses linear prediction to achieve noise reduction in a voice signal which has been mixed with an ambient noise (Signal to Noise (S-N) ratio = about 0dB) is proposed. This noise reduction method which is based on the linear prediction estimates the voice spectrum while ignoring the spectrum of the noise. The performance of the noise reduction method is first examined using the transversal linear predictor filter. However, with this method there is deterioration in the tone quality of the predicted voice due to the low level of the S-N ratio. An additional processing circuit is then proposed so as to adjust the noise reduction circuit with an aim of improving the problem of tone deterioration. Next, we consider a practical application where the effects of round on errors arising from fixed-point computation has to be minimized. This minimization is achieved by using the lattice predictor filter which in comparison to the transversal type, is Down to be less sensitive to the round-off error associated with finite word length operations. Finally, we consider a practical application where noise reduction is necessary. In this noise reduction method, both the voice spectrum and the actual noise spectrum are estimated. Noise reduction is achieved by using the linear predictor filter which includes the control of the predictor filter coefficient’s update.

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A study on speech enhancement using complex-valued spectrum employing Feature map Dependent attention gate (특징 맵 중요도 기반 어텐션을 적용한 복소 스펙트럼 기반 음성 향상에 관한 연구)

  • Jaehee Jung;Wooil Kim
    • The Journal of the Acoustical Society of Korea
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    • v.42 no.6
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    • pp.544-551
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    • 2023
  • Speech enhancement used to improve the perceptual quality and intelligibility of noise speech has been studied as a method using a complex-valued spectrum that can improve both magnitude and phase in a method using a magnitude spectrum. In this paper, a study was conducted on how to apply attention mechanism to complex-valued spectrum-based speech enhancement systems to further improve the intelligibility and quality of noise speech. The attention is performed based on additive attention and allows the attention weight to be calculated in consideration of the complex-valued spectrum. In addition, the global average pooling was used to consider the importance of the feature map. Complex-valued spectrum-based speech enhancement was performed based on the Deep Complex U-Net (DCUNET) model, and additive attention was conducted based on the proposed method in the Attention U-Net model. The results of the experiments on noise speech in a living room environment showed that the proposed method is improved performance over the baseline model according to evaluation metrics such as Source to Distortion Ratio (SDR), Perceptual Evaluation of Speech Quality (PESQ), and Short Time Object Intelligence (STOI), and consistently improved performance across various background noise environments and low Signal-to-Noise Ratio (SNR) conditions. Through this, the proposed speech enhancement system demonstrated its effectiveness in improving the intelligibility and quality of noisy speech.

Adaptive Noise Canceller by Weight Updating Control Method for Speech Enhancement (음성향상을 위한 가중치 갱신제어방식의 적응소음제거기)

  • Kim, Gyu-Dong;Lee, Yun-Jung;Kim, Pil-Un;Chang, Yong-Min;Cho, Jin-Ho;Kim, Myoung-Nam
    • Journal of Korea Multimedia Society
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    • v.10 no.8
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    • pp.1004-1016
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    • 2007
  • In this paper we proposed a Weight-Update-Control Adaptive Noise Canceller which improves speech when environmental noise is stationary and it is hard to acquire a reference signal. Adaptive Noise Canceller(ANC) needs a reference signal, but it is not easy to measure pure noise without voice for reference in factory. Because there are mixed various mechanical noise and workers' voice. Therefore ANC is not suitable to reduce background noise. So we proposed the method that uses an arbitrary constant as an input signal and inputs microphone signal to the reference signal. The noise is eliminated using updated weights in non-speech range. In speech range the weight is fixed and the modified voice is acquired then voice is restored through transversal filter. The proposed method is based on facts that the factory noise is stationary and the noise is not changed in short conversation range. As a result of simulation using MATLAB, we confirmed that the proposed method is effective for reducing factory noise and has high signal to noise ratio(SNR).

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Prediction of Performance Loss Due to Phase Noise in Digital Satellite Communication System (디지털 위성통신시스템에서 위상 잡음으로 인한 성능 손실 예측)

  • 김영완;박동철
    • The Journal of Korean Institute of Electromagnetic Engineering and Science
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    • v.13 no.7
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    • pp.679-686
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    • 2002
  • Based on the alternating series expansion of error probability function due to phase noise in PSK systems, the performance evaluations for Tikhonov and Gaussian probability density functions were performed in this paper. The range of the signal-to-noise ratio of recovered carrier signal which provides the same dependency between the error performances by Tikhonov function and Gaussian function was analyzed via loss evaluation due to phase noise. The phase noise with 1/f$^2$ characteristic was generated based on the relationship of the phase noise spectral density and the modulation index for frequency modulation signal. Using the generated phase noise as the input signal for digital satellite communication receiver, the performance losses due to the phase noise were measured and evaluated with the analyzed performance characteristics.

A Study on the Scaling in Wave Digital Filter (웨이브 디지털 필터의 스케일링에 관한 연구)

  • 권희훈;김명기
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.12 no.1
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    • pp.27-35
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    • 1987
  • Digital filter suffer from roundoff noise and adder overflows due to finite word length effects. Scaling is an attempt to internal signal levels such that all signals are as large as possible, yet without the occurrence of overflows. Scaling requirements are implemented by the use of transformer. This paper proposes a procedure for scaling wave digital filters to avoid overflow problems and at the same time maximizing the output signal-to-noise ratio. Results indicate that the scaled networks have an improved signal to noise ratio over th unscaled filters under the condition that there be no overflow occuring.

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Determination of the Optimum-Bandwidth of Chirp-Signal for Pulse Compression Technique (펄스압축 기술을 위한 chirp 신호의 최적대역폭 결정)

  • Ko, Dae-Sik;Moon, Gun
    • The Journal of the Acoustical Society of Korea
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    • v.16 no.2
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    • pp.5-9
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    • 1997
  • In this paper, when we use the chirp signal as input signal of ultrasonic signal system the technique for determining the bandwidth of the chirp signal that maximizes the amplitude of the compressed ultrasonic echo signal has been studied. In ultrasonic signal processing systems, the signal-to-noise ratio of the echo signal can be too low due to damping and scattering of the ultrasonic wave during transmission. Method of pulse compression using chirp signal is a means to increase the signal-to-noise ratio in ultrasonic pulse-echo systems. Simulation and experimental results showed that the output signal of ultrasonic system was increased by pulse width of chirp signal and the optimum-bandwidth of the chirp signal was 1.15 times larger than the bandwidth of the ultrasonic system.

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Partial Discharge Signal Denoising using Adaptive Translation Invariant Wavelet Transform-Online Measurement

  • Maheswari, R.V.;Subburaj, P.;Vigneshwaran, B.;Iruthayarajan, M. Willjuice
    • Journal of Electrical Engineering and Technology
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    • v.9 no.2
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    • pp.695-706
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    • 2014
  • Partial discharge (PD) measurements have emerged as a dominant investigative tool for condition monitoring of insulation in high voltage equipment. But the major problem behind them the PD signal is severely polluted by several noises like White noise, Random noise, Discrete Spectral Interferences (DSI) and the challenge lies with removing these noise from the onsite PD data effectively which leads to preserving the signal for feature extraction. Accordingly the paper is mainly classified into two parts. In first part the PD signal is artificially simulated and mixed with white noise. In second part the PD is measured then it is subjected to the proposed denoising techniques namely Translation Invariant Wavelet Transform (TIWT). The proposed TIWT method remains the edge of the original signal efficiently. Additionally TIWT based denoising is used to suppress Pseudo Gibbs phenomenon. In this paper an attempt has been made to review the methodology of denoising the PD signals and shows that the proposed denoising method results are better when compared to other wavelet-based approaches like Fast Fourier Transform (FFT), Discrete Wavelet Transform (DWT), by evaluating five different parameters like, Signal to noise ratio, Cross-correlation coefficient, Pulse amplitude distortion, Mean square error, Reduction in noise level.

Compensation for the decrease of output SNR of hadamard transform spectrometer with nonideal mask (비이상적 마스크로 인한 하다마드변환 스펙트럼 검파기 출력값의 신호대 잡음비 감소의 해결방안)

  • 남지탁;박진배;윤태성
    • 제어로봇시스템학회:학술대회논문집
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    • 1997.10a
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    • pp.4-7
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    • 1997
  • When we use Hadamard transform spectrometers (HTS), we can increase signal to noise ratio(SNR) by multiplexing which is done by masks. But if the mask has a single defective element, output-SNR decreases. In this paper the effect of a single defective element on the output-SNR is investigated. And a method of compensating for the defective mask element is presented.

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Two-Microphone Generalized Sidelobe Canceller with Post-Filter Based Speech Enhancement in Composite Noise

  • Park, Jinsoo;Kim, Wooil;Han, David K.;Ko, Hanseok
    • ETRI Journal
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    • v.38 no.2
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    • pp.366-375
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    • 2016
  • This paper describes an algorithm to suppress composite noise in a two-microphone speech enhancement system for robust hands-free speech communication. The proposed algorithm has four stages. The first stage estimates the power spectral density of the residual stationary noise, which is based on the detection of nonstationary signal-dominant time-frequency bins (TFBs) at the generalized sidelobe canceller output. Second, speech-dominant TFBs are identified among the previously detected nonstationary signal-dominant TFBs, and power spectral densities of speech and residual nonstationary noise are estimated. In the final stage, the bin-wise output signal-to-noise ratio is obtained with these power estimates and a Wiener post-filter is constructed to attenuate the residual noise. Compared to the conventional beamforming and post-filter algorithms, the proposed speech enhancement algorithm shows significant performance improvement in terms of perceptual evaluation of speech quality.

Research on the optimization method for PGNAA system design based on Signal-to-Noise Ratio evaluation

  • Li, JiaTong;Jia, WenBao;Hei, DaQian;Yao, Zeen;Cheng, Can
    • Nuclear Engineering and Technology
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    • v.54 no.6
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    • pp.2221-2229
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    • 2022
  • In this research, for improving the measurement performance of Prompt Gamma-ray Neutron Activation Analysis (PGNAA) set-up, a new optimization method for set-up design was proposed and investigated. At first, the calculation method for Signal-to-Noise Ratio (SNR) was proposed. Since the SNR could be calculated and quantified accurately, the SNR was chosen as the evaluation parameter in the new optimization method. For discussing the feasibility of the SNR optimization method, two kinds of PGNAA set-ups were designed in the MCNP code, based on the SNR optimization method and the previous signal optimization method, respectively. Meanwhile, the single element spectra analysis method was proposed, and the analysis effect of single element spectra as well as element sensitivity were used for comparing the measurement performance. Since the simulation results showed the better measurement performance of set-up designed by SNR optimization method, the experimental set-ups were built for the further testing, finally demonstrating the feasibility of the SNR optimization method for PGNAA setup design.