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    Wyświetlanie 1-5 z 5
    Tytuł:
    Recognizing the Sequences of Code Manipulated Short LFM Signals
    Autorzy:
    Leszczyński, T.
    Powiązania:
    https://bibliotekanauki.pl/articles/176689.pdf
    Data publikacji:
    2012
    Wydawca:
    Polska Akademia Nauk. Czytelnia Czasopism PAN
    Tematy:
    sonar
    LFM signal
    matched filtering
    Opis:
    Noise-like binary sequences combined with signals with linear frequency modulation might be suc- cessfully used to increase the reliability of the recognition of both probe and communication signals in the presence of natural and artificial interference. To identify such formed sequences the usage of the two-step matched filtering was suggested and the probabilistic model of the recognition of noise-like code sequences transferred by LFM signals was developed.
    Źródło:
    Archives of Acoustics; 2012, 37, 3; 295-300
    0137-5075
    Pojawia się w:
    Archives of Acoustics
    Dostawca treści:
    Biblioteka Nauki
    Artykuł
    Tytuł:
    Optimization of Short Probe Linear Frequency Modulated Signal Parameters
    Autorzy:
    Pogribny, W.
    Leszczyński, T.
    Powiązania:
    https://bibliotekanauki.pl/articles/176953.pdf
    Data publikacji:
    2011
    Wydawca:
    Polska Akademia Nauk. Czytelnia Czasopism PAN
    Tematy:
    sonar
    LFM signal
    matched filtering
    Opis:
    In many physical experiments, linear frequency modulated (LFM) signals are widely used to probe objects in different environments, from outer-space to under- water. These signals allow a significant improvement in measurement resolution, even when the observation distance is great. For example, using LFM probe signals in underwater investigations enables discovery of even small objects covered by bottom sediments. Recognition of LFM (chirp) signals depends on their compression based on matched filtering. This work presents two simple solutions to improve the resolution of the short chirp signals recognition. These methods are effective only if synchronization between the signal and matched filter (MF) is obtained. This work describes both the aforementioned methods and a method of minimizing the effects of the lack of synchronization. The proposed matched filtering method, with the use of n parallel MFs and other techniques, allows only one sample to be obtained in the main lobe and to accurately locate its position in the appropriate sampling period Ts with accuracy Ts/n. These approaches are appropriate for use in probe signal processing.
    Źródło:
    Archives of Acoustics; 2011, 36, 4; 861-871
    0137-5075
    Pojawia się w:
    Archives of Acoustics
    Dostawca treści:
    Biblioteka Nauki
    Artykuł
    Tytuł:
    Study on the Effectiveness of Monte Carlo Filtering when Correcting Negative SEA Loss Factors
    Autorzy:
    Nieradka, Paweł
    Dobrucki, Andrzej
    Powiązania:
    https://bibliotekanauki.pl/articles/31339865.pdf
    Data publikacji:
    2023
    Wydawca:
    Polska Akademia Nauk. Czasopisma i Monografie PAN
    Tematy:
    statistical energy analysis
    coupling loss factor
    Monte Carlo filtering
    power injection method
    Opis:
    The power injection method (PIM) is an experimental method used to identify the statistical energy analysis (SEA) parameters (called loss factors – LFs) of a vibroacoustic system. By definition, LFs are positive real numbers. However, it is not uncommon to obtain negative LFs during experiments, which is considered a measurement error. To date, a recently proposed method, called Monte Carlo filtering (MCF), of correcting negative coupling loss factors (CLFs) has been validated for systems that meet SEA assumptions. In this article, MCF was validated for point connections and in conditions where SEA assumptions are not met (systems with low modal overlap, non-conservative junctions, strong coupling). The effect of removing MCF bias on the results was also examined. During the experiments, it was observed that the bias is inversely proportional to the damping loss factor of the examined subsystems. The obtained results confirm that the PIM, combined with MCF, allows to determine non-negative SEA parameters in all considered cases.
    Źródło:
    Archives of Acoustics; 2023, 48, 3; 201-218
    0137-5075
    Pojawia się w:
    Archives of Acoustics
    Dostawca treści:
    Biblioteka Nauki
    Artykuł
    Tytuł:
    Deep Neural Network for Supervised Single-Channel Speech Enhancement
    Autorzy:
    Saleem, Nasir
    Irfan Khattak, Muhammad
    Ali, Muhammad Yousaf
    Shafi, Muhammad
    Powiązania:
    https://bibliotekanauki.pl/articles/177497.pdf
    Data publikacji:
    2019
    Wydawca:
    Polska Akademia Nauk. Czasopisma i Monografie PAN
    Tematy:
    deep neural network
    intelligibility
    speech enhancement
    speech quality
    supervised learning
    Wiener filtering
    Opis:
    Speech enhancement is fundamental for various real time speech applications and it is a challenging task in the case of a single channel because practically only one data channel is available. We have proposed a supervised single channel speech enhancement algorithm in this paper based on a deep neural network (DNN) and less aggressive Wiener filtering as additional DNN layer. During the training stage the network learns and predicts the magnitude spectrums of the clean and noise signals from input noisy speech acoustic features. Relative spectral transform-perceptual linear prediction (RASTA-PLP) is used in the proposed method to extract the acoustic features at the frame level. Autoregressive moving average (ARMA) filter is applied to smooth the temporal curves of extracted features. The trained network predicts the coefficients to construct a ratio mask based on mean square error (MSE) objective cost function. The less aggressive Wiener filter is placed as an additional layer on the top of a DNN to produce an enhanced magnitude spectrum. Finally, the noisy speech phase is used to reconstruct the enhanced speech. The experimental results demonstrate that the proposed DNN framework with less aggressive Wiener filtering outperforms the competing speech enhancement methods in terms of the speech quality and intelligibility.
    Źródło:
    Archives of Acoustics; 2019, 44, 1; 3-12
    0137-5075
    Pojawia się w:
    Archives of Acoustics
    Dostawca treści:
    Biblioteka Nauki
    Artykuł
    Tytuł:
    Stability Conditions for the Leaky LMS Algorithm Based on Control Theory Analysis
    Autorzy:
    Bismor, D.
    Pawelczyk, M.
    Powiązania:
    https://bibliotekanauki.pl/articles/177088.pdf
    Data publikacji:
    2016
    Wydawca:
    Polska Akademia Nauk. Czytelnia Czasopism PAN
    Tematy:
    adaptive filtering
    leaky LMS
    stability
    negative step size
    identification
    adaptive line enhancer
    active noise control
    Opis:
    The Least Mean Square (LMS) algorithm and its variants are currently the most frequently used adaptation algorithms; therefore, it is desirable to understand them thoroughly from both theoretical and practical points of view. One of the main aspects studied in the literature is the influence of the step size on stability or convergence of LMS-based algorithms. Different publications provide different stability upper bounds, but a lower bound is always set to zero. However, they are mostly based on statistical analysis. In this paper we show, by means of control theoretic analysis confirmed by simulations, that for the leaky LMS algorithm, a small negative step size is allowed. Moreover, the control theoretic approach alows to minimize the number of assumptions necessary to prove the new condition. Thus, although a positive step size is fully justified for practical applications since it reduces the mean-square error, knowledge about an allowed small negative step size is important from a cognitive point of view.
    Źródło:
    Archives of Acoustics; 2016, 41, 4; 731-739
    0137-5075
    Pojawia się w:
    Archives of Acoustics
    Dostawca treści:
    Biblioteka Nauki
    Artykuł
      Wyświetlanie 1-5 z 5

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