A method and an apparatus for filtering an audio signal include the steps of making the audio signal available in digitalized form, wherein the duration of the sampling interval is half or less than half of the period duration of the highest frequency to be expected in the audio signal, a digitalized impulse response is made available in accordance with the desired filtering effect, and a convolution sum is formed from the samples of the impulse response and the samples of the audio signal, wherein, (i) several adjacent samples of the impulse response define a corresponding interval within at least a time portion of the impulse response which is shorter than the impulse response, (ii) within the interval defined in this manner, the samples of the impulse response corresponding to the interval defined in this manner are equated to a value which is a function of one or more of the samples of the digitalized impulse response falling within the interval defined in this manner, and (iii) the steps (i) and (ii) are repeated as necessary, with the requirement that the intervals defined in this manner do not overlap, so that for computing the convolution sum a time-coarsened impulse response is utilized for the convolution at least in one time portion, and the impulse response is otherwise used unchanged for the convolution.
The invention relates to a method for calculating a least mean square algorithm using an N-tap filter. A modified least mean square algorithm is used for the calculation. In a first calculation step, calculation is effected using a first, wide bit width in the N taps. In a second step, depending on the result of the first calculation step, a second, smaller number of bits is selected for the further method steps.
The invention provides a method and system for forming an output impulse response function. The method includes the steps of creating an initial impulse response, and dividing the impulse response into a head portion and a tail portion. The tail portion is high pass filtered, and low frequency components of the head portion are boosted. The low frequency boosted and high pass filtered respective head and tail portions are then combined into a modified output impulse response, which can then be used to spatialize an audio signal by convolving it.
A method and apparatus for applying a gain characteristic to an audio signal are provided. Data storing a plurality of gain characteristics at a plurality of different levels is stored in a storage means. The amplitude of an input signal is repeatedly assessed and from this a gain characteristic to be applied to the input is determined.
