1. A method for processing an audio signal, the method comprising: filtering audio frames of the audio signal using the following filter: H(z)=(1−B(z))/(1−C(z)·z −T ) where B(z) and C(z) are two FIR filters and the H(z) filter parameters are the coefficients of the FIR filters B(z) and C(z), and T indicates the pitch lag, wherein filtering the audio frames of the audio signal comprises filtering a past frame of the audio signal using a set of first filter parameters for obtaining a filtered past frame, and filtering a current frame of the audio signal using a set of second filter parameters for obtaining a filtered current frame, and removing a discontinuity between the filtered past frame and the filtered current frame of the audio signal using a linear predictive filter, wherein initial states of the linear predictive filter are obtained by filtering a last part of the past frame using the set of second filter parameters.
2. The method of claim 1 , further comprising estimating the linear predictive filter using the audio signal.
3. The method of claim 2 , wherein estimating the linear predictive filter comprises estimating the linear predictive filter based on the past and/or current frame of the audio signal or based on the filtered past frame of the audio signal using the Levinson-Durbin algorithm.
4. The method of claim 1 , wherein the linear predictive filter comprises a linear predictive filter of an audio codec.
5. A non-transitory digital storage medium having stored thereon a computer program product for performing, when run on a computer, a method for processing an audio signal, the method comprising: filtering audio frames of the audio signal using the following filter: H(z)=(1−B(z))/(1−C(z)·z −T ) where B(z) and C(z) are two FIR filters and the H(z) filter parameters are the coefficients of the FIR filters B(z) and C(z), and T indicates the pitch lag, wherein filtering the audio frames of the audio signal comprises filtering a past frame of the audio signal using a set of first filter parameters for obtaining a filtered past frame, and filtering a current frame of the audio signal using a set of second filter parameters for obtaining a filtered current frame, and removing a discontinuity between the filtered past frame and the filtered current frame of the audio signal using a linear predictive filter, wherein initial states of the linear predictive filter are obtained by filtering a last part of the past frame using the set of second filter parameters.
6. An apparatus for processing an audio signal, wherein the apparatus comprises a processor configured to: filter audio frames of the audio signal using the following filter: H(z)=(1−B(z))/(1−C(z)·z −T ) where B(z) and C(z) are two FIR filters and the H(z) filter parameters are the coefficients of the FIR filters B(z) and C(z), and T indicates the pitch lag, wherein filtering the audio frames of the audio signal comprises filtering a past frame of the audio signal using a set of first filter parameters for obtaining a filtered past frame, and filtering a current frame of the audio signal using a set of second filter parameters for obtaining a filtered current frame, and remove a discontinuity between the filtered past frame and the filtered current frame of the audio signal using a linear predictive filter, wherein initial states of the linear predictive filter are obtained by filtering a last part of the past frame using the set of second filter parameters.
7. An audio decoder, comprising an apparatus of claim 6 .
8. An audio encoder, comprising an apparatus of claim 6 .
