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ABSTRACT: Recently, a variable step-size affine projection (VSS-AP) algorithm has been introduced. The algorithm provides faster convergence rate and lower misadjustment but has heavy computational complexity. Proposed is a new variable step-size pseudo affine projection (VSS-PAP) algorithm, which not only has a much lower complexity than the VSS-AP algorithm but also provides performance comparable with the conventional algorithm. Simulation results confirm fast convergence rate and small misalignment of the proposed algorithm with less computational complexity.
Electronics Letters 02/2008; · 0.96 Impact Factor
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ABSTRACT: An efficient method for controlling the echo in a continuously variable slope delta modulation (CVSD) mobile communication channel is proposed. By utilising the symmetry of the coder pairs, the proposed method shortens the echo path length and reduces the nonlinear distortion inherent in the coders. An additional advantage for the VLSI implementation is obtained using the combined bit-stream. The proposed method shows an improved echo return loss enhancement (ERLE) performance and reduced complexity compared to the conventional method.
Electronics Letters 06/2000; · 0.96 Impact Factor
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ABSTRACT: The filtered-x LMS algorithm and its modified versions have been successfully applied in suppressing acoustic noise such as single and multiple tones and broadband random noise. This paper presents an adaptive algorithm based on the filtered-x LMS algorithm which may be applied in attenuating tonal acoustic noise. In the proposed method, the weights of the adaptive filter and estimation of the phase shift due to the acoustic path from a loudspeaker to a microphone are computed simultaneously for optimal control. The algorithm possesses advantages over other filtered-x LMS approaches in three aspects: (1) each frequency component is processed separately using an adaptive filter with two coefficients, (2) the convergence parameter for each sinusoid can be selected independently, and (3) the computational load can be reduced by eliminating the convolution process required to obtain the filtered reference signal. Simulation results for a single-input/single-output (SISO) environment demonstrate that the proposed method is robust to the changes of the acoustic path between the actuator and the microphone and outperforms the filtered-x LMS algorithm in simplicity and convergence speed.
The Journal of the Acoustical Society of America 08/1998; 104(1):248-54. · 1.55 Impact Factor
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ABSTRACT: A fast vector-sum codebook search method for low bit rate speech
coding is presented. In this method, the codebook search is simplified
by designing a vector-sum codebook that consists of orthonormal regular
pulse basis vectors. A further simplification is achieved by adopting
backward filtering. The method proposed has significantly reduced
computational complexity, compared with the conventional VSELP, without
producing any additional degradation in the quality of the synthesised
speech
Electronics Letters 04/1997; · 0.96 Impact Factor
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ABSTRACT: The authors present an adaptive eigenfilter implementing the
optimum moving target indication (MTI) processor. The method presented
shows a fast convergence capability as well as better steady-state
performance than the prediction error filter (PEF). Moreover, the
arithmetic efficiency of the method results from using modular
processing architecture which takes advantage of the Gram-Schmidt (GS)
processor
Electronics Letters 09/1996; · 0.96 Impact Factor
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ABSTRACT: The authors derive expressions of the output
signal-to-interference plus noise ratio (SINR) of the linearly
constrained narrowband beamformer in noncoherent situations using a
vector approach. Numerical results are included
IEEE Transactions on Antennas and Propagation 11/1993; · 2.15 Impact Factor
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ABSTRACT: The authors propose an adaptive processor implementing an optimum moving target indicator (MTI) in the sense of maximizing the improvement factor (IF). A multichannel scheme for preventing coefficient fluctuation due to the presence of target echoes or impulse interferences is also discussed. The proposed eigenfilter recursively estimates the minimum eigenvector of the input clutter covariance matrix based on the inverse power method (IPM). In the proposed scheme, the IPM iteration is performed in a numerically efficient way using systolic processing architectures. Simulation results demonstrate that the convergence speed of the eigenfilter is faster than that of the prediction error filters and IF values of the eigenfilter always approach the optimum. The multichannel adaptive eigenfilter is shown to be a robust scheme preventing coefficient fluctuation in the presence of target echoes
Acoustics, Speech, and Signal Processing, 1993. ICASSP-93., 1993 IEEE International Conference on; 05/1993
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ABSTRACT: An efficient algorithm detecting the presence of a fetal QRS complex is presented. The proposed fetal QRS detection method computes the averaged magnitude of the difference between the fetal ECG signal and the reference signal to detect the fetal QRS event. The detected fetal QRS complexes are exponentially averaged to generate the template signal which can track the slowly varying shape of the fetal ECG signal. As an effort to obtain improved detection performances, two approaches of normalizing the fetal ECG signal and the template are considered.
IEEE Transactions on Biomedical Engineering 09/1992; 39(8):868-71. · 2.28 Impact Factor
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ABSTRACT: An adaptive digital filter structure which can be used in
multichannel noise-canceling applications is described. The proposed
structure is obtained by creating a hybrid from the lattice and the
escalator (Gram-Schmidt) structures, with the coefficients being updated
using the least mean square (LMS) algorithm. As an application, the
proposed adaptive multichannel digital filter is applied to implement an
adaptive generalized sidelobe canceler
IEEE Transactions on Signal Processing 08/1992; · 2.63 Impact Factor
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ABSTRACT: The convergence rate for the adaptive weights to reach the optimum
value in an adaptive array system depends on the eigenvalue spread ratio
of the autocovariance matrix. How the eigenvalue spread ratio in a
generalized sidelobe canceller (GSC) is affected by the various
parameters is studied. Expressions for the output power of the GSC in
coherent situations are derived, and numerical results are
included
IEEE Transactions on Antennas and Propagation 05/1992; · 2.15 Impact Factor
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ABSTRACT: An adaptive multichannel digital filter structure that can be used
in multichannel noise-cancelling applications is described. The proposed
structure is a hybrid of the lattice and the escalator (Gram-Schmidt)
structures, and the coefficients are updated using the least-mean-square
algorithm. In the lattice-escalator hybrid structure, complete
decorrelation of multiple reference signals is achieved by successively
generating forward prediction error vectors via the lattice structure
and decorrelating the elements of the error vectors using escalator
linear combiners at each lattice stage. Simulation results indicate that
the convergence speed of the hybrid structure is as fast as the
escalator realization
Acoustics, Speech, and Signal Processing, 1990. ICASSP-90., 1990 International Conference on; 05/1990 · 4.63 Impact Factor
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ABSTRACT: The authors describe the implementation of a real-time fetal ECG
(electrocardiogram) monitoring system in which an adaptive multichannel
noise canceller is realized using the TMS32020 programmable digital
signal processor. An ECG signal from the electrode placed on the
mother's abdomen and three ECGs from those on the chest are applied as
the desired signal and the reference inputs, respectively, of the
multichannel filter. The coefficients of the filter are updated using
the LMS (least-mean-square) algorithm, such that the output of the
multichannel filter copies the maternal ECG embedded in the abdominal
ECG. The enhanced fetal ECG is obtained by subtracting the filter output
from the abdominal ECG, and the difference signal is recorded
Circuits and Systems, 1988., IEEE International Symposium on; 07/1988
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ABSTRACT: The authors update delta modulation digital filter weights using
the LMS (least-mean-squares) and the SIGN algorithms to realize an
adaptive digital filter without multiplication operations. It is shown
that using the SIGN algorithm results in an adaptive filter that can be
implemented using the simple up/down counting operations. Learning
curves demonstrating convergence properties of the algorithms for a
system identification problem are presented
Proceedings of the IEEE 06/1988; · 6.81 Impact Factor
