Time-frequency domain adaptive filters
A linear system representation called the double convolution model, which combines the impulse response and transfer function, is presented. It is based on Fourier transformation by block of the impulse response. Its main advantage is that the model order and the window size are not imposed by the response length. This property makes the adaptation of systems with long impulse response easier because it may be possible to use a moderate order and to avoid large processing delay. Simulation results are presented for an acoustic channel, the impulse response of which is measured in a reverberant room.<<ETX>>