“…Audio‐conferencing is a crucial application that demands adaptive filters with large numbers of coefficients and fast convergence rate in order to properly avoid perceptible acoustic echo. These requirements are stringent even for state‐of‐the‐art adaptive filtering (AF) algorithms [1]. It is widely known that colored input signals (which are common in practice) reduce the convergence rate of AFs, and subband adaptive filtering (SAF) algorithms have been proposed to mitigate such issue.…”
Speech enhancement and acoustic noise reduction are two important tasks where adaptive filtering algorithms emerge as a competitive solution. Unfortunately, in such applications the convergence rate of the system identification is hampered when the excitation data is highly correlated. Subband adaptive algorithms have been developed to address such issue. A recently proposed low‐complexity subband adaptive structure with sparse subfilters is generalised, in order to permit a non‐uniform filter bank structure. This generalisation brings additional flexibility to the resulting critically decimated adaptive structure, which allows one to adapt it to the idiosyncrasies of the application of interest without losing the perfect‐reconstruction property. A closed‐form solution for the optimal values that the adaptive coefficient should assume in order to accurately emulate a given impulse response is derived. Simulations reveal that the resulting algorithm may outperform recently published subband adaptive filtering algorithms, thereby requiring even less computational effort.
“…Audio‐conferencing is a crucial application that demands adaptive filters with large numbers of coefficients and fast convergence rate in order to properly avoid perceptible acoustic echo. These requirements are stringent even for state‐of‐the‐art adaptive filtering (AF) algorithms [1]. It is widely known that colored input signals (which are common in practice) reduce the convergence rate of AFs, and subband adaptive filtering (SAF) algorithms have been proposed to mitigate such issue.…”
Speech enhancement and acoustic noise reduction are two important tasks where adaptive filtering algorithms emerge as a competitive solution. Unfortunately, in such applications the convergence rate of the system identification is hampered when the excitation data is highly correlated. Subband adaptive algorithms have been developed to address such issue. A recently proposed low‐complexity subband adaptive structure with sparse subfilters is generalised, in order to permit a non‐uniform filter bank structure. This generalisation brings additional flexibility to the resulting critically decimated adaptive structure, which allows one to adapt it to the idiosyncrasies of the application of interest without losing the perfect‐reconstruction property. A closed‐form solution for the optimal values that the adaptive coefficient should assume in order to accurately emulate a given impulse response is derived. Simulations reveal that the resulting algorithm may outperform recently published subband adaptive filtering algorithms, thereby requiring even less computational effort.
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