The invention relates to a method for processing a signal, in particular a digital audio signal, suitable for being implemented by a digital signal processor (DSP) having libraries for calculating Fourier transforms from the complex number space to the complex number space, for digitally processing P input signals, P being an integer at least equal to 2, more particularly for filtering said P input signals by the convolution of sampled fast Fourier transforms (FFT), thus obtaining Q output signals, Q being an integer at least equal to 2. According to the invention, the method includes at least the following steps: —grouping said P input signals by twos, one representing the real portion and the other the imaginary portion of a complex number, thus defining one or more input vectors, —filtering said input vector or vectors, passing through the Fourier space, thus generating one or more output vectors, which are complex numbers, the real portion and the imaginary portion of said vector or each one of said output vectors respectively representing one of said Q output signals.