• ETRI Journal
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• 제38권2호
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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.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2003년도 하계종합학술대회 논문집 Ⅳ
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• pp.1711-1714
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• 2003

The noise in an image degrades image quality and deteriorates coding efficiency of compression. Recently, various edge-preserving noise filtering methods based on the nonstationary image model have been proposed to overcome this problem. In most conventional nonstationary image models, however, pixels are assumed to be uncorrelated to each other In order not to increase the computational burden too much. As a result, some detailed information is lost in the filtered results. In this paper, we propose a computationally feasible adaptive noise smoothing algorithm which considers the nonstationary correlation characteristics of images. We assume that an image has a nonstationary mean and can be segmented into subimages which have individually different stationary correlations. Taking advantage of the special structure of the covariance matrix that results from the proposed image model, we derive a computationally efficient FFT-based adaptive linear minimum mean square error filter. The justification for the proposed image model is presented and the effectiveness of the proposed algorithm is demonstrated experimentally.

• International Journal of Control, Automation, and Systems
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• 제6권6호
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• pp.818-827
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• 2008

In this paper, we propose a new noise estimation and reduction algorithm for stationary and nonstationary noisy environments. This approach uses an algorithm that classifies the speech and noise signal contributions in time-frequency bins. It relies on the ratio of the normalized standard deviation of the noisy power spectrum in time-frequency bins to its average. If the ratio is greater than an adaptive estimator, speech is considered to be present. The propose method uses an auto control parameter for an adaptive estimator to work well in highly nonstationary noisy environments. The auto control parameter is controlled by a linear function using a posteriori signal to noise ratio(SNR) according to the increase or the decrease of the noise level. The estimated clean speech power spectrum is obtained by a modified gain function and the updated noisy power spectrum of the time-frequency bin. This new algorithm has the advantages of much more simplicity and light computational load for estimating the stationary and nonstationary noise environments. The proposed algorithm is superior to conventional methods. To evaluate the algorithm's performance, we test it using the NOIZEUS database, and use the segment signal-to-noise ratio(SNR) and ITU-T P.835 as evaluation criteria.

• 한국소음진동공학회논문집
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• 제12권8호
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• pp.589-597
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• 2002

주행중인 차량의 동적 거동에 대한 응답을 해석하기 위하여 시간영역에서 뿐만 아니라 주파수 영역에서의 해석이 필요하다. 주파수 영역에서의 해석을 위하여 시간영역에서의 값을 FFT를 이용하여 주파수 영역으로 변화하는 방법이 일반적으로 사용되어 왔다. 본 연구에서는 최대 엔트로피 방법을 이용하여 기존의 FFT 방법보다 차량의 과도응답특성을 시간영역 및 주파수 영역에서 편리하게 해석할 수 있는 방법을 제시하고 있다.

• 한국음향학회지
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• 제21권1호
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• pp.11-18
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• 2002

본 논문에서는 부가 잡음에 오염된 음성신호에 이득 적응을 가지는 음성인식을 시간 영역에서 다루었다. 잡음은 유색잡음이라고 가정한다. 전화망에서 마찰음 (fricative), 운음 (glides), 유음 (liquds), 그리고 천이영역(transition region)과 같은 음성 신호의 뚜렷한 비정상상태를 극복하기 위해서 NAR-HMM (nonstationary autoregressive HMM)7을 제안하였다. 비정상상태 AR 처리는 M개의 알고 있는 기저 함수 (basis function)의 선형 결합으로 이루어진 다항 함수 (polynomial function)로 나타낼 수 있다. 오염된 신호만을 이용할 수 있을 때, 잡음의 추정 (estimation)문제는 필연적으로 발생한다. 다중 Kalman 필터를 사용함으로써, 잡음모델의 추정과 음성의 이득곡선 (gain contour)을 수행하였다. 제안한방법의 잡음 추정은 오염된 신호로부터 효과적으로 잡음을 제거하여 깨끗한 음성신호를 얻을 수 있었다. 또한 잡음 추정을 하는 일반적인 ARHMM보다 제안한 NAR-HMM이 약 2-3%의 인식성능을 향상시켰다.

• 한국음향학회:학술대회논문집
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• 한국음향학회 1998년도 제15회 음성통신 및 신호처리 워크샵(KSCSP 98 15권1호)
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• pp.225-230
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• 1998

In this paper, a Kalman filter approach for enhancing speech signals degraded by statistically independent additive nonstationary noise is developed. The autoregressive hidden markov model is used for modeling the statistical characteristics of both the clean speech signal and the nonstationary noise process. In this case, the speech enhancement comprises a weighted sum of conditional mean estimators for the composite states of the models for the speech and noise, where the weights equal to the posterior probabilities of the composite states, given the noisy speech. The conditional mean estimators use a smoothing spproach based on two Kalmean filters with Markovian switching coefficients, where one of the filters propagates in the forward-time direction with one frame. The proposed method is tested against the noisy speech signals degraded by Gaussian colored noise or nonstationary noise at various input signal-to-noise ratios. An app개ximate improvement of 4.7-5.2 dB is SNR is achieved at input SNR 10 and 15 dB. Also, in a comparison of conventional and the proposed methods, an improvement of the about 0.3 dB in SNR is obtained with our proposed method.

• 한국소음진동공학회논문집
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• 제22권3호
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• pp.291-297
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• 2012

Conventional Fourier analysis can give only limited information about the dynamic characteristics of nonstationary signals. Instead, time-frequency analysis is widely used to investigate the nonstationary signal in detail. Several time-frequency analysis methods are compared for a typical acoustic emission burst generated during the impact between a ferrite ceramic and aluminum plate. This AE burst is inherently nonstationary and random containing many frequency contents, which leads to severe interference between cross terms in bilinear convolution type distributions. The smoothing and reassignment processes can improve the readability and resolution of the results. Spectrogram and scalogram of the AE burst are obtained and compared to get the characteristics information. Renyi entropies are computed for various bilinear time-frequency transforms to evaluate the randomness. These bilinear transforms are reassigned by using the improved algorithm in discrete computation.

• 한국소음진동공학회:학술대회논문집
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• 한국소음진동공학회 1997년도 추계학술대회논문집; 한국과학기술회관; 6 Nov. 1997
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• pp.345-351
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• 1997

The motion of an aircraft landing gear over rough runway at variable speed is nonstationary. hi this paper, a method for the computation of nonstationary response variance is presented which uses a state space form for the combination of landing gear and runway excitation. The dynamic characteristics of the landing gear under nonstationazy random excitations has also been analyzed using the proposed method. The formulation is for linear systems of arbitrary order and allows any deterministic velocity history.

• 한국음향학회지
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• 제34권4호
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• pp.310-315
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• 2015

본 논문에서는 비정체성(nonstationary) 잡음 환경을 위한 단일 채널 VAD(Voice Activity Detection) 알고리듬 제안하였다. VAD 판별을 위한 특징계수의 임계값은 과거 비음성 프레임들의 평균과 표준편차를 추산하여 적응적으로 갱신하였다. 특징계수로는 SPD-TE(Spectral Power Difference-Teager Energy)를 사용했는데, 이것은 WPD(Wavelet Packet Decomposition) 계수에 Teager 에너지를 적용한 것으로서 잡음에 강인한 것으로 보고된 바 있다. TIMIT 음성과 NOISEX-92 잡음을 사용하여 10 dB부터 -10 dB까지의 SNR에 대한 실험 결과, 제안된 알고리듬이 표준을 포함한 기존의 알고리듬과 비슷한 정확도를 보였다.

• 한국음향학회지
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• 제27권7호
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• pp.386-393
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• 2008

본 논문에서는 비정상 잡음환경에서 음질향상을 위한 새로운 방법을 제안한다. 정상 잡음환경에서 음질향상을 위한 잡음제거 방법으로 주파수 차감법이 잘 알려져 있다. 그러나 실제 잡음환경은 대 부분 비정상적인 특성을 나타낸다. 제안한 방법은 다양한 잡음 과 비정상 환경에서 잘 동작 할 수 있도록 적응 임계 치를 위한 자동제어 파라미터를 사용한다. 특히, 자동제어 파라미터는 a posteriori SNR을 이용한 선형함수를 적용하여 잡음레벨의 증감에 따라 적응 임계 치를 제어한다. 제안한 알고리즘은 음질향상을 위해 Hangover (HO)을 이용한 주파수 차감법과 결합한다. 알고리즘의 성능은 다양한 잡음환경에서 ITU-T P.835 signal distortion (SIG)와 segment signal to-noise ratio (SNR)로 평가하여 (HO)을 이용한 음성검출과 minimum statistics (MS) 방법에 비해 우수한 결과를 나타냈다