Description and generation of spherically invariant speech-model signals
Signal Processing
Digital signal processing (3rd ed.): principles, algorithms, and applications
Digital signal processing (3rd ed.): principles, algorithms, and applications
Adaptive filter theory (3rd ed.)
Adaptive filter theory (3rd ed.)
Step-size control for acoustic echo cancellation filter—an overview
Signal Processing - Special issue on current topics in adaptive filtering for hands-free acoustic communication and beyond
Signal Processing - Special issue on current topics in adaptive filtering for hands-free acoustic communication and beyond
Adaptation of a memoryless preprocessor for nonlinear acoustic echo cancelling
Signal Processing - Special issue on current topics in adaptive filtering for hands-free acoustic communication and beyond
The generalized multidelay adaptive filter: structure andconvergence analysis
IEEE Transactions on Signal Processing
Baseband Volterra filters for implementing carrier basednonlinearities
IEEE Transactions on Signal Processing
Adaptive nonlinear system identification in the short-time fourier transform domain
IEEE Transactions on Signal Processing
IEEE Transactions on Signal Processing
Modeling and identification of nonlinear systems in the short-time fourier transform domain
IEEE Transactions on Signal Processing
On the performance of adaptive pruned Volterra filters
Signal Processing
Generalized Spherical Array Beamforming for Binaural Speech Reproduction
IEEE/ACM Transactions on Audio, Speech and Language Processing (TASLP)
| Hi-index | 0.09 |
Standard approaches for the cancellation of acoustic echoes in telecommunication systems assume that the echo path to be identified can be modeled by a linear system. However, in applications such as echo cancellation for mobile communication terminals, non-negligible nonlinear distortions are introduced by loudspeakers and their amplifiers, resulting in a significant degradation of the performance of purely linear approaches. In this contribution we consider so-called power filters in order to cope with these kinds of nonlinear echo paths. By combining time-domain orthogonalization methods with a DFT-domain implementation of power filters we derive corresponding quasi-complete orthogonalized versions. As the statistical properties of speech input are non-stationary, the orthogonalization must follow this time-variance, too. Furthermore, we introduce a step-size control for the adaptation of the equivalent orthogonalized structure. Experiments with real hardware show that, with the proposed nonlinear approach, an increase in echo attenuation can be achieved, if the loudspeaker system introduces saturation-type nonlinearities in the echo path.