Filters








14 Hits in 4.2 sec

Subband adaptive filtering for acoustic echo control using allpass polyphase IIR filterbanks

P.A. Naylor, O. Tanrikulu, A.G. Constantinides
1998 IEEE Transactions on Speech and Audio Processing  
The scheme, in conjunction with normalized least mean squares (NLMS) adaptive filters, is tested in an acoustic echo control application and shown to give better convergence, lower delay, and lower computational  ...  The scheme is based on infinite impulse response (IIR) filterbanks, formed from allpass polyphase filters, which exhibit very high quality filtering compared to typical finite impulse response (FIR) implementations  ...  ACKNOWLEDGMENT The authors wish to thank all the partners of CEC ESPRIT Project 6166 (FREETEL) for their helpful collaboration, and in particular the Speech Processing Group, MATRA Communication, France  ... 
doi:10.1109/89.661473 fatcat:vprl7vmcfvemzfjev3v5xuvi3m

Rfi Cancellation In Vdsl Systems Using A Novel Complex

Fernando Hugo Gregorio, Juan Edmundo Cousseau, T.I. Laakso
2005 Zenodo  
Publication in the conference proceedings of EUSIPCO, Antalya, Turkey, 2005  ...  An allpass-based IIR multiple Complex Adaptive Notch Filter (CANF) is presented in this work, with two different algorithms, associated to the Recursive Prediction Error (RPE) method (A1-CANF) and an alternative  ...  NOVEL ALLPASS-BASED CANF (A-CANF) The model for the proposed allpass-based multiple complex adaptive notch filter (A-CANF) can be described by H(z) = 1 2 (1 + M ∏ i=1 V i (z)) (7) where V i (z) = ρ i −  ... 
doi:10.5281/zenodo.38944 fatcat:cf55zonbmbcy3oglyfavh47ime

Adaptive Filtering Using Subband Processing: Application to Background Noise Cancellation [chapter]

Ali O. Abid Noor, Salina Abdul, Aini Hussai
2011 Adaptive Filtering  
The new scheme is based on using polyphase allpass IIR filter banks at the analysis stage, while the synthesis filter bank is optimized such that an inherent phase correction is made at the output of the  ...  A possible solution to reduce the complexity problem has been to use adaptive Infinite Impulse Response (IIR) filters, such that an effectively long impulse response can be achieved with relatively few  ...  In this section, an adaptive noise cancellation scheme that uses a combination of polyphase allpass filter banks at the analysis stage and an optimized FIR filter bank at the synthesis stage is developed  ... 
doi:10.5772/16363 fatcat:zd2mgxuznrfhzdvmlzbhskytra

A New Adaptive Notch Filtering Algorithm Based on Normalized Lattice Structure with Improved Mean Update Term

Shinichiro NAKAMURA, Shunsuke KOSHITA, Masahide ABE, Masayuki KAWAMATA
2015 IEICE Transactions on Fundamentals of Electronics Communications and Computer Sciences  
SUMMARY In this paper, we propose Affine Combination Lattice Algorithm (ACLA) as a new lattice-based adaptive notch filtering algorithm.  ...  Furthermore, we show that this step-size bound is characterized by the gradient of the mean update term. key words: adaptive notch filter, affine combination, mean update term, normalized lattice structure  ...  Among many types of adaptive notch filters, we focus on the one based on the secondorder IIR notch filter that is designed by an IIR all-pass filter [1] , [4] - [6] , [9] - [13] , [15] , [18] .  ... 
doi:10.1587/transfun.e98.a.1482 fatcat:sex6kbpg3bdpxm6i7ispe3hruq

Recent Advances in Variable Digital Filters [chapter]

Shunsuke Koshita, Masahide Abe, Masayuki Kawamata
2018 Digital Systems  
In the topic on adaptive filtering, we introduce the details of adaptive bandpass/band-stop filtering that include the well-known adaptive notch filtering.  ...  Variable digital filters are widely used in a number of applications of signal processing because of their capability of self-tuning frequency characteristics such as the cutoff frequency and the bandwidth  ...  In [32] [33] [34] , the normalized lattice structure is applied to construct the notch filter, and the adaptive algorithm makes use of the state variable of the normalized lattice structure instead of  ... 
doi:10.5772/intechopen.79198 fatcat:3isab2wi4fdhtmjm7aybvrut7q

Channel Equalization in Filter Bank Based Multicarrier Modulation for Wireless Communications

Tero Ihalainen, Tobias Hidalgo Stitz, Mika Rinne, Markku Renfors
2006 EURASIP Journal on Advances in Signal Processing  
A novel structure, consisting of a linear-phase FIR amplitude equalizer and an allpass filter as phase equalizer, is found to provide enhanced robustness to timing estimation errors.  ...  We utilize an efficient oversampled filter bank concept with 2x-oversampled subcarrier signals that can be equalized independently of each other.  ...  In this case, each subchannel equalizer comprises a cascade of a first-order complex allpass filter, a phase rotator combined with the operation of taking the real part of the signal, and a first-order  ... 
doi:10.1155/2007/49389 fatcat:4bk6rotkcngbddnh5dlj3pvcke

2010 Index IEEE Transactions on Signal Processing Vol. 58

2010 IEEE Transactions on Signal Processing  
., +, TSP Nov. 2010 5682-5692 Steady-State MSE Performance Analysis of Mixture Approaches to Adaptive Filtering. Kozat, S.  ...  Avargel, Y., +, TSP Dec. 2010 6052-6065 Notch filters Generalized Adaptive Notch Smoothing Revisited.  ...  Global Positioning System A Fixed-Lag Particle Filter for the Joint Detection/Compensation of Interference Effects in GPS Navigation.  ... 
doi:10.1109/tsp.2010.2092533 fatcat:4y66ezuo7zf6doe6nwjqwtc42i

Table of contents

2004 2004 IEEE International Symposium on Circuits and Systems (IEEE Cat No 04CH37512) ISCAS-04  
... 265 METHOD -IIR WHITENING FILTERING APPROACH Jonah Gamba, Yusuke Tsuda, Tetsuya Shimamura, Saitama University, Japan DSP-L8.4: MSE ANALYSIS OF AN ALLPASS FILTER-BASED ADAPTIVE IIR NOTCH FILTER .  ...  III -269 WITH A NORMALIZED ALGORITHM Aloys Mvuma, Hiroshima University, Japan; Shotaro Nishimura, Shimane University, Japan; Takao Hinamoto, Hiroshima University, Japan DSP-L8.5: ROBUST ADAPTIVE BEAMFORMING  ... 
doi:10.1109/iscas.2004.1328114 fatcat:xmvsmkhxgbb55ftuygqomthikm

Symmetric convolution using unitary transform matrices

T.M. Foltz, B.M. Welsh, C.D. Holmberg
2000 IEEE Transactions on Signal Processing  
Abstract-In this correspondence, the fast algorithms to calculate a sequence of adaptive Cosine-Walsh transforms aimed for the compression of multidimensional nuclear data are defined.  ...  All formulae are mainly derived for symmetric and antimetric wave digital filters having either an even or an odd characteristic function.  ...  Ford of Air Force Research Lab, Kirtland AFB, NM, for their support of this research project.  ... 
doi:10.1109/78.863086 fatcat:5ljirlexvnfyldgsyubpn6kpya

Perceptual coding of digital audio

T. Painter, A. Spanias
2000 Proceedings of the IEEE  
Next, filter bank design issues and algorithms are addressed, with a particular emphasis placed on the modified discrete cosine transform, a perfect reconstruction cosine-modulated filter bank that has  ...  This paper concludes with a discussion of future research directions.  ...  In the study, two IIR and two FIR coding schemes were constructed from the template using a structured all-pass filter bank, a parallel allpass filter bank, a tree-structured QMF bank, and a polyphase  ... 
doi:10.1109/5.842996 fatcat:jkfvoxg7zrcyxg6fyahe4u73pu

Orthonormal bases for adaptive filtering [article]

HJW Harm Belt, HJ Butterweck, WMG Wim Van Bokhoven, AC Bert Den Brinker
1997
The ALE input d(k) is assumed to be the sum of a narrow-band random signal s(k) •This chapter is based on a paper presented in cooperation with A.C. den Brinker and F.P.A.  ...  The Second-Order IIR Prediction Filter 127 In this chapter an ALE based on a second-order IIR filter is proposed that includes adaptation of the filter bandwidth.  ...  The adaptive LRM can, under some circumstances, suffice with much less adaptive parameters than an adaptive FIR filter. This chapter deals with the choice of the set of (IIR) filters in an LRM .  ... 
doi:10.6100/ir491853 fatcat:47di6uhxnfbrboqehff7ihrdaq

REVIEW: AUDIO NOISE REDUCTION USING FILTERS AND DISCRETE WAVELET TRANSFORMATION

Er Kaur, Shalu Rani
2016 unpublished
Such as SNR, PSNR, MSE and the Time to reduce the noise for noisy signals for removing noise.  ...  Therefore, basic linear Filters are used to denoise the audio signals and enhance speech and audio signal quality.  ...  Based on the number of samples at a stretch that are corrupted, an adaptive filter with a variable size window is applied on the corrupted speech signal to remove the impulse noise.  ... 
fatcat:75zezy32lnajpovsoswbzda35y

Compact and accurate hardware simulation of wireless channels for single and multiple antenna systems

Saeed Fouladi Fard
2009
I am glad that I had the opportunity of knowing and collaborating with Amirhossein Alimohammad.  ...  I had a good time with my friend and they inspired me and made me feel home at  ...  His maturity, support, and enthusiasm helped me grow and gave me courage to say no to "the Spirit of Gravity." His insightful help and guidance significantly improved this work in many aspects.  ... 
doi:10.7939/r3rm7j fatcat:mwv5dzxiobfubeama3zr5ur2fa

Local sound field synthesis

Fiete Winter
2019
In agreement with numerical sound field simulations, a specifically developed geometric model shows an increase of synthesis accuracy for LSFS compared to conventional SFS approaches.  ...  of timbral fidelity for distinct parametrisations.  ...  Since the inverse of an allpass generally results in an unstable filter, the so-called backward filtering approach is utilised: The inverse of an allpass filter is equivalent to its conjugate complex,  ... 
doi:10.18453/rosdok_id00002568 fatcat:5zerlwzdjrbcbpzyuqsetopqjm