• Title/Summary/Keyword: non-stationary 배경잡음

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Non-Stationary/Mixed Noise Estimation Algorithm Based on Minimum Statistics and Codebook Driven Short-Term Predictor Parameter Estimation (최소 통계법과 Short-Term 예측계수 코드북을 이용한 Non-Stationary/Mixed 배경잡음 추정 기법)

  • Lee, Myeong-Seok;Noh, Myung-Hoon;Park, Sung-Joo;Lee, Seok-Pil;Kim, Moo-Young
    • The Journal of the Acoustical Society of Korea
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    • v.29 no.3
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    • pp.200-208
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    • 2010
  • In this work, the minimum statistics (MS) algorithm is combined with the codebook driven short-term predictor parameter estimation (CDSTP) to design a speech enhancement algorithm that is robust against various background noise environments. The MS algorithm functions well for the stationary noise but relatively not for the non-stationary noise. The CDSTP works efficiently for the non-stationary noise, but not for the noise that was not considered in the training stage. Thus, we propose to combine CDSTP and MS. Compared with the single use of MS and CDSTP, the proposed method produces better perceptual evaluation of speech quality (PESQ) score, and especially works excellent for the mixed background noise between stationary and non-stationary noises.

Background Noise Classification in Noisy Speech of Short Time Duration Using Improved Speech Parameter (개량된 음성매개변수를 사용한 지속시간이 짧은 잡음음성 중의 배경잡음 분류)

  • Choi, Jae-Seung
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.20 no.9
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    • pp.1673-1678
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    • 2016
  • In the area of the speech recognition processing, background noises are caused the incorrect response to the speech input, therefore the speech recognition rates are decreased by the background noises. Accordingly, a more high level noise processing techniques are required since these kinds of noise countermeasures are not simple. Therefore, this paper proposes an algorithm to distinguish between the stationary background noises or non-stationary background noises and the speech signal having short time duration in the noisy environments. The proposed algorithm uses the characteristic parameter of the improved speech signal as an important measure in order to distinguish different types of the background noises and the speech signals. Next, this algorithm estimates various kinds of the background noises using a multi-layer perceptron neural network. In this experiment, it was experimentally clear the estimation of the background noises and the speech signals.

Robust Speech Enhancement Based on Soft Decision Employing Spectral Deviation (스펙트럼 변이를 이용한 Soft Decision 기반의 음성향상 기법)

  • Choi, Jae-Hun;Chang, Joon-Hyuk;Kim, Nam-Soo
    • Journal of the Institute of Electronics Engineers of Korea SP
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    • v.47 no.5
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    • pp.222-228
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    • 2010
  • In this paper, we propose a new approach to noise estimation incorporating spectral deviation with soft decision scheme to enhance the intelligibility of the degraded speech signal in non-stationary noisy environments. Since the conventional noise estimation technique based on soft decision scheme estimates and updates the noise power spectrum using a fixed smoothing parameter which was assumed in stationary noisy environments, it is difficult to obtain the robust estimates of noise power spectrum in non-stationary noisy environments that spectral characteristics of noise signal such as restaurant constantly change. In this paper, once we first classify the stationary noise and non-stationary noise environments based on the analysis of spectral deviation of noise signal, we adaptively estimate and update the noise power spectrum according to the classified noise types. The performances of the proposed algorithm are evaluated by ITU-T P. 862 perceptual evaluation of speech quality (PESQ) under various ambient noise environments and show better performances compared with the conventional method.

Speech Enhancement System Using a Model of Auditory Mechanism (청각기강의 모델을 이용한 음성강조 시스템)

  • 최재승
    • Journal of the Institute of Electronics Engineers of Korea SP
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    • v.41 no.6
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    • pp.295-302
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    • 2004
  • On the field of speech processing the treatment of noise is still important problems for speech research. Especially, it has been noticed that the background noise causes remarkable reduction of speech recognition ratio. As the examples of the background noise, there are such various non-stationary noises existing in the real environment as driving noise of automobiles on the road or typing noise of printer. The treatment for these kinds of noises is not so simple as could be eliminated by the former Wiener filter, but needs more skillful techniques. In this paper as one of these trials, we show an algorithm which is a speech enhancement method using a model of mutual inhibition for noise reduction in speech which is contaminated by white noise or background noise mentioned above. It is confirmed that the proposed algorithm is effective for the speech degraded not only by white noise but also by colored noise, judging from the spectral distortion measurement.

Speech Enhancement Based on Modified IMCRA Using Spectral Minima Tracking with Weighted Subband Selection (서브밴드 가중치를 적용한 스펙트럼 최소값 추적을 이용하는 수정된 IMCRA 기반의 음성 향상 기법)

  • Park, Yun-Sik;Park, Gyu-Seok;Lee, Sang-Min
    • Journal of the Institute of Electronics Engineers of Korea SP
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    • v.49 no.3
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    • pp.89-97
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    • 2012
  • In this paper, we propose a novel approach to noise power estimation for speech enhancement in noisy environments. The method based on IMCRA (improved minima controlled recursive averaging) which is widely used in speech enhancement utilizes a rough VAD (voice activity detection) algorithm which excludes speech components during speech periods in order to improves the performance of the noise power estimation by reducing the speech distortion caused by the conventional algorithm based on the minimum power spectrum derived from the noisy speech. However, since the VAD algorithm is not sufficient to distinguish speech from noise at non-stationary noise and low SNRs (signal-to-noise ratios), the speech distortion resulted from the minimum tracking during speech periods still remained. In the proposed method, minimum power estimate obtained by IMCRA is modified by SMT (spectral minima tracking) to reduce the speech distortion derived from the bias of the estimated minimum power. In addition, in order to effectively estimate minimum power by considering the distribution characteristic of the speech and noise spectrum, the presented method combines the minimum estimates provided by IMCRA and SMT depending on the weighting factor based on the subband. Performance of the proposed algorithm is evaluated by subjective and objective quality tests under various environments and better results compared with the conventional method are obtained.