摘要:
The present invention relates to a method and an apparatus for processing a signal, which are used to effectively reproduce an audio signal, and more particularly, to a method and an apparatus for processing an audio signal, which are used for implementing a filtering for input audio signals with a low computational complexity. To this end, provided are a method for processing an audio signal including: receiving an input audio signal; receiving truncated subband filter coefficients for filtering each subband signal of the input audio signal, the truncated subband filter coefficients being at least a portion of subband filter coefficients obtained from binaural room impulse response (BRIR) filter coefficients for binaural filtering of the input audio signal, the lengths of the truncated subband filter coefficients being determined based on filter order information obtained by at least partially using characteristic information extracted from the corresponding subband filter coefficients, and the truncated subband filter coefficients being constituted by at least one FFT filter coefficient in which fast Fourier transform (FFT) by a predetermined block size in the corresponding subband has been performed; performing the fast Fourier transform of the subband signal based on a predetermined subframe size in the corresponding subband; generating a filtered subframe by multiplying the fast Fourier transformed subframe and the FFT filter coefficients; inverse fast Fourier transforming the filtered subframe; and generating a filtered subband signal by overlap-adding at least one subframe which is inverse fast Fourier transformed and an apparatus for processing an audio signal using the same.
摘要:
The present invention relates to a method and an apparatus for processing a signal, which are used to effectively reproduce an audio signal, and more particularly, to a method for generating a filter for an audio signal, which are used for implementing a filtering for input audio signals with a low computational complexity and a parameterization apparatus therefor. To this end, provided are a method for generating a filter of an audio signal, including: receiving at least one proto-type filter coefficient for filtering each subband signal of an input audio signal; converting the proto-type filter coefficient into a plurality of subband filter coefficients; truncating each of the subband filter coefficients based on filter order information obtained by at least partially using characteristic information extracted from the corresponding subband filter coefficients, the length of at least one truncated subband filter coefficients being different from the length of truncated subband filter coefficients of another subband; and generating FFT filter coefficients by fast Fourier transforming (FFT) the truncated subband filter coefficients by a predetermined block size in the corresponding subband and a parameterization unit using the same.