Informacja

Drogi użytkowniku, aplikacja do prawidłowego działania wymaga obsługi JavaScript. Proszę włącz obsługę JavaScript w Twojej przeglądarce.

Wyszukujesz frazę "filtracja adaptacyjna" wg kryterium: Temat


Wyświetlanie 1-4 z 4
Tytuł:
Quality assessment of speech signals under a process of echo cancelation in telecommunications systems
Autorzy:
Dobrucki, Andrzej
Kin, Maurycy
Walczyński, Maciej
Powiązania:
https://bibliotekanauki.pl/articles/2146665.pdf
Data publikacji:
2022
Wydawca:
Politechnika Poznańska. Instytut Mechaniki Stosowanej
Tematy:
DSP
digital signal processing
echo cancellation
adaptive filtration
sound perception
cyfrowe przetwarzanie sygnałów
kompensacja echa
filtracja adaptacyjna
percepcja dźwięku
Opis:
The phenomenon of echo in the telecommunications channels is caused by the reflection of an electrical signal in a long line. In order to improve the quality of the transmitted sound, various adaptive filters are used to remove or at least reduce the level of the reflected delayed signal. However, such a process may result in a degradation in the quality of speech, although its intelligibility may not get worse. The work presents the results of subjective studies on assessing the quality of speech signals under the process of acoustic echo cancellation using different algorithms. The algorithms studied were: LMS (Least Mean Squares), NLMS (Normalized Least Mean Squares) and AP (Affine Projection). The study consisted of assessing the signal quality after applying the echo elimination process using the Degradation Category Rating method. A total of 312 signals were used in the test: 192 male speech and 120 female speech samples. Echo simulation was used using different delay times and levels of echo signal. Both types of speech have signal delay times of 20 ms, 50 ms, 100 ms and 200 ms with echo level values of -6 dB, -12 dB, -18 dB and -24 dB. In addition, for female speech signals, a delay time of 150 ms was introduced. The study involved 14 people aged between 18 and 38, including six women and eight men. All subjects had normal good hearing. Seven listeners had participated in subjective listening tests of the sound quality assessment previously. The listeners’ opinions were collected on prepared questionnaire. It was found that the highest ratings were given to the AP filter, while the worst ratings were featured the NLMS. It should also be noted that the range between the results obtained for AP and NLMS for female speech is smaller in comparison to male. It is also interesting that the discrepancy in ratings was greatest for a delay time of 100 ms for the AP filter and 200 ms for the LMS filter. It can therefore be concluded from the obtained results that, in the case of acoustic echo cancelation, AP filter introduced the lowest quality degradation while the LMS achieved slightly worse average ratings when compared to the AP filter . The NLMS filter characterized by the worst ratings, and in some cases received twice the quality degradation compared to the AP filter.
Źródło:
Vibrations in Physical Systems; 2022, 33, 1; art. no. 2022111
0860-6897
Pojawia się w:
Vibrations in Physical Systems
Dostawca treści:
Biblioteka Nauki
Artykuł
Tytuł:
Sequential separation of twin pregnancy electrocardiograms
Autorzy:
Kotas, M.
Leski, J. M.
Wrobel, J.
Powiązania:
https://bibliotekanauki.pl/articles/202099.pdf
Data publikacji:
2016
Wydawca:
Polska Akademia Nauk. Czytelnia Czasopism PAN
Tematy:
fetal ECG
twin pregnancy
ECG signals decomposition
blind source separation
independent component analysis
source subspaces
projective filtering
adaptive filtering
EKG płodu
ciąża bliźniacza
EKG
separacja sygnałów niewidomych
niezależna analiza składowych
podprzestrzenie źródła
filtracja adaptacyjna
Opis:
We propose to tackle the problem of maternal abdominal electric signals decomposition with a combined application of independent component analysis and projective or adaptive filtering. The developed method is employed to process the four-channel abdominal signals recorded during twin pregnancy. These signals are complicated mixtures of the maternal ECG, the ECGs of the fetal twins and noise of various origin. Although the independent component analysis cannot separate the respective signals, the proposed combination of the methods deals with this task successfully. A simulation experiment confirms high efficiency of this approach.
Źródło:
Bulletin of the Polish Academy of Sciences. Technical Sciences; 2016, 64, 1; 91-101
0239-7528
Pojawia się w:
Bulletin of the Polish Academy of Sciences. Technical Sciences
Dostawca treści:
Biblioteka Nauki
Artykuł
Tytuł:
Automatic speech signal segmentation based on the innovation adaptive filter
Autorzy:
Makowski, R.
Hossa, R.
Powiązania:
https://bibliotekanauki.pl/articles/330096.pdf
Data publikacji:
2014
Wydawca:
Uniwersytet Zielonogórski. Oficyna Wydawnicza
Tematy:
automatic speech segmentation
inter phoneme boundaries
Schur adaptive filtering
detection threshold determination
automatyczna segmentacja mowy
filtracja adaptacyjna
określenie progu detekcji
Opis:
Speech segmentation is an essential stage in designing automatic speech recognition systems and one can find several algorithms proposed in the literature. It is a difficult problem, as speech is immensely variable. The aim of the authors’ studies was to design an algorithm that could be employed at the stage of automatic speech recognition. This would make it possible to avoid some problems related to speech signal parametrization. Posing the problem in such a way requires the algorithm to be capable of working in real time. The only such algorithm was proposed by Tyagi et al., (2006), and it is a modified version of Brandt’s algorithm. The article presents a new algorithm for unsupervised automatic speech signal segmentation. It performs segmentation without access to information about the phonetic content of the utterances, relying exclusively on second-order statistics of a speech signal. The starting point for the proposed method is time-varying Schur coefficients of an innovation adaptive filter. The Schur algorithm is known to be fast, precise, stable and capable of rapidly tracking changes in second order signal statistics. A transfer from one phoneme to another in the speech signal always indicates a change in signal statistics caused by vocal track changes. In order to allow for the properties of human hearing, detection of inter-phoneme boundaries is performed based on statistics defined on the mel spectrum determined from the reflection coefficients. The paper presents the structure of the algorithm, defines its properties, lists parameter values, describes detection efficiency results, and compares them with those for another algorithm. The obtained segmentation results, are satisfactory.
Źródło:
International Journal of Applied Mathematics and Computer Science; 2014, 24, 2; 259-270
1641-876X
2083-8492
Pojawia się w:
International Journal of Applied Mathematics and Computer Science
Dostawca treści:
Biblioteka Nauki
Artykuł
Tytuł:
Metoda filtracji sygnału fonokardiograficznego
Denoising method for heart sounds
Autorzy:
Wilk, B.
Powiązania:
https://bibliotekanauki.pl/articles/156082.pdf
Data publikacji:
2011
Wydawca:
Stowarzyszenie Inżynierów i Techników Mechaników Polskich
Tematy:
sygnał FKG
adaptacyjna filtracja zakłóceń
analiza falkowa
heart sounds
adaptive noise cancellation
wavelet analysis
Opis:
Zakłócenia zewnętrzne (tj. sygnały akustyczne pochodzące z otoczenia), które występują podczas rejestracji tonów i szmerów serca (tzw. sygnału FKG) za pomocą stetoskopu z mikrofonem, znacznie utrudniają analizę zjawisk osłuchowych. W artykule przedstawiono metodę adaptacyjnej filtracji zakłóceń zewnętrznych. Jako sygnał wejściowy filtru LMS przyjęto aproksymację sygnału FKG uzyskaną na podstawie analizy falkowej zakłóconego FKG. Natomiast na wejście referencyjne filtru LMS podano zmodyfikowany sygnał reprezentujący zakłócenia zewnętrzne zarejestrowane drugim mikrofonem. Proponowana metoda daje lepsze rezultaty niż filtracja adaptacyjna przeprowadzona bezpośrednio na zarejestrowanych sygnałach.
Auscultation is one of the most important non-invasive and simple diagnostic tools for detecting disorders of cardiac diseases. The intelligent stethoscope with microphone allows us to acquire and analyze heart sounds (produced by the mechanical action of the heart) objectively. In daily life various noise components, especially from a noisy environment, make the diagnostic evaluation of a phonocardiographic signal (i.e. FCG) difficult or in some cases even impossible. Heart sounds and additive external noises usually overlap in the frequency domain. This paper presents a novel wavelet-based adaptive denoising method for external noise cancellation from the FCG signal. Heart sounds and acoustic ambient noises were recorded separately by two microphones at the same time (Fig. 1). At first, the recorded signals are decomposed by DWT using "coif5" wavelet. The sum of the details at levels from 5 up to 8 gives an acceptable approximation of the FCG signal (Fig. 3) and it was used as the primary input of the LMS adaptive filter (Fig. 2). The reference signal required for the LMS filter was derived from the modified wavelet coefficients of the acquired noise signal (Fig. 5). This signal is highly correlated with artifacts in the FCG signal and it provides better results of adaptive noise cancellation than the raw signal corresponding to acoustic ambient noise. The comparison between the power of the FCG signal segments, including, and void of, external noise, and the power of the filtered signal can be used to measure the filtering effectiveness. The results show that the proposed method is a promising technique for external noise reduction in the FCG signal.
Źródło:
Pomiary Automatyka Kontrola; 2011, R. 57, nr 12, 12; 1495-1497
0032-4140
Pojawia się w:
Pomiary Automatyka Kontrola
Dostawca treści:
Biblioteka Nauki
Artykuł
    Wyświetlanie 1-4 z 4

    Ta witryna wykorzystuje pliki cookies do przechowywania informacji na Twoim komputerze. Pliki cookies stosujemy w celu świadczenia usług na najwyższym poziomie, w tym w sposób dostosowany do indywidualnych potrzeb. Korzystanie z witryny bez zmiany ustawień dotyczących cookies oznacza, że będą one zamieszczane w Twoim komputerze. W każdym momencie możesz dokonać zmiany ustawień dotyczących cookies