“…A very useful tool to express the effect of echo cancellation is the Echo Return Loss Enhancement (ERLE) [12] defined as:…”
Section: Discussion and Analysis Of Simulation Resultsmentioning
confidence: 99%
“…For each iteration the LMS algorithm requires 2N additions and 2N+1 multiplications (N for calculating the output, y(n), one for 2μe(n) and an additional N for the scalar by vector multiplication) [11], [12], [17]. Figure 5 shows the flowchart of the basic LMS adaptive filtering Algorithm.…”
The Conventional acoustic echo canceller encounters problems like slow convergence rate (especially for speech signal) and high computational complexity as the identification of the echo path requires filter with more than a thousand taps, non-stationary speech input, slowly timevarying systems to be identified. The demand for fast convergence and less MSE level cannot be met by conventional adaptive filtering algorithms. There is a need to be computationally efficient and rapidly converging algorithm.The LMS algorithm is easy to implement and computationally inexpensive. This feature makes the LMS algorithm attractive for echo cancellation applications. The results show that the steady state value of the output estimation error increases with increasing the step size parameter and the optimality of the LMS algorithm is no longer hold. The results also reveal that choosing the smallest value of the step size parameter guarantees the smallest mis-adjustment but might not meet the convergence criteria.
General TermsAdaptive Filtering Algorithm, Acoustic Echo-cancellation.
“…A very useful tool to express the effect of echo cancellation is the Echo Return Loss Enhancement (ERLE) [12] defined as:…”
Section: Discussion and Analysis Of Simulation Resultsmentioning
confidence: 99%
“…For each iteration the LMS algorithm requires 2N additions and 2N+1 multiplications (N for calculating the output, y(n), one for 2μe(n) and an additional N for the scalar by vector multiplication) [11], [12], [17]. Figure 5 shows the flowchart of the basic LMS adaptive filtering Algorithm.…”
The Conventional acoustic echo canceller encounters problems like slow convergence rate (especially for speech signal) and high computational complexity as the identification of the echo path requires filter with more than a thousand taps, non-stationary speech input, slowly timevarying systems to be identified. The demand for fast convergence and less MSE level cannot be met by conventional adaptive filtering algorithms. There is a need to be computationally efficient and rapidly converging algorithm.The LMS algorithm is easy to implement and computationally inexpensive. This feature makes the LMS algorithm attractive for echo cancellation applications. The results show that the steady state value of the output estimation error increases with increasing the step size parameter and the optimality of the LMS algorithm is no longer hold. The results also reveal that choosing the smallest value of the step size parameter guarantees the smallest mis-adjustment but might not meet the convergence criteria.
General TermsAdaptive Filtering Algorithm, Acoustic Echo-cancellation.
“…Reduced size predictors in the FTF algorithms have been successfully used in the FNTF algothms [7,8]. The SMFTF algorithm can easily used with reduced size prediction part (table 2).…”
Section: Simplified Ftf-type Algorithmmentioning
confidence: 99%
“…In this application, predictor sizes are much smaller than the size of the transversal filter for speech signal. This propriety was used to develop a class of algorithms called Fast Newton transversal filter algorithm [7,8] where the input signal is modelised by an AR model with 10 to 20 coefficients. From relation (6), we can see that the most significant components, the last ones, of the backward predictor affect the last terms of the Kalman Gain and this contribution is not In the proposed algorithm, we discard all backward prediction variables from (6) and use only the forward variables to compute the dual Kalman gain :…”
“…However, for medium-sized "lters (about 250 taps as discussed below) such as those encountered in mobile handsfree applications, FRLS algorithms still have high complexity. To overcome this problem, the use of a class of Newton-type algorithms known as fast Newton transversal "lters (FNTF) has been proposed for mobile applications [55]. These algorithms have proved to be particularly well suited to speech since they assume low-order AR models for the input signals; moreover they exhibit better robustness to background noise than LMS-type algorithms.…”
This paper gives an overview of recent bibliographic references dealing with speech processing in mobile terminals. Its purpose is to point out state of the art issues in the area; thus a fairly large list of references taken from many conferences proceedings and journals is given and commented. General considerations about speech processing in mobile communications are "rstly introduced; then we deal with audio processing for speech enhancement in mobile terminals and with low bit-rate speech coding. Speech recognition is addressed with some accent put on mobile applications. A short overview of implementation aspects of speech processing algorithms in mobile terminals is also given. Finally, open issues and problems are listed.2000 Published by Elsevier Science B.V. All rights reserved.
ZusammenfassungDiese Arbeit gibt einen U G berblick auf neue bibliogra"sche Referenzen, die sich mit der Sprachverarbeitung in mobilen Einheiuten beschaK ftigen. Unser Ziel ist es, den Stand der technischen VeroK !entlichungen auf diesem Gebiet zu ermitteln; somit ist eine recht lange Liste von Referenzen aus vielen Konferenzen und Zeitschiften und die zugehoK rigen Kommentare angegeben. Allgemeine U G berlegungen zur Sprachverarbeitung in der Mobilkommunikation werden als erstes angefuK hrt; anschlie{end behandeln wir Verfahren zur Spracheverbesserung in mobilen Systemen basierend auf niedrigen Bitraten. Bei der anschlieend behandelten Spracherkennung wird ein gewisser Schwerpunkt auf mobile Systeme gelegt. Ein kurzer U G berblick zum Implementierungsaspekt der Sparchverarbeitungsverfahren in mobilen EndgeraK ten ist ebenfalls angegeben. Schlielich werden o!ene Themen und Probleme aufgezeigt.2000 Published by Elsevier Science B.V. All rights reserved.
Re2 sume2Cet article passe en revue la bibliographie reH cente concernant le traitement de la parole dans les terminaux mobiles. Son objectif est de mettre l'accent sur l'eH tat de l'art dans le domaine; en conseH quence une importante liste de reH feH rences recueillies dans des actes de congre`s et des revues est donneH e et commenteH e. En premier lieu, des consideH rations geH neH rales sur le traitement de la parole dans les communications mobiles sont preH senteH es; puis nous discutons de traitements de rehaussement de la parole dans les terminaux mobiles ainsi que du codage de la parole a`bas deH bits. La reconnaissance vocale est traiteH e avec un accent particulier mis sur les applications mobiles. L'implantation des algorithmes de traitement de la parole dans les terminaux mobiles est aussi brie`vement traiteH e. En"n, certaines questions et proble`mes ouverts sont citeH s.
scite is a Brooklyn-based organization that helps researchers better discover and understand research articles through Smart Citations–citations that display the context of the citation and describe whether the article provides supporting or contrasting evidence. scite is used by students and researchers from around the world and is funded in part by the National Science Foundation and the National Institute on Drug Abuse of the National Institutes of Health.