Method for generating filter for audio signal, and parameterization device for same
The present invention relates to a method for generating a filter for an audio signal and a parameterization device for the same, and more particularly, to a method for generating a filter for an audio signal, to implement filtering of an input audio signal with a low computational complexity, and a parameterization device therefor. To this end, provided are a method for generating a filter for an audio signal, including: receiving at least one binaural room impulse response (BRIR) filter coefficients for binaural filtering of an input audio signal; converting the BRIR filter coefficients into a plurality of subband filter coefficients; obtaining average reverberation time information of a corresponding subband by using reverberation time information extracted from the subband filter coefficients; obtaining at least one coefficient for curve fitting of the obtained average reverberation time information; obtaining flag information indicating whether the length of the BRIR filter coefficients in a time domain is more than a predetermined value; obtaining filter order information for determining a truncation length of the subband filter coefficients, the filter order information being obtained by using the average reverberation time information or the at least one coefficient according to the obtained flag information and the filter order information of at least one subband being different from filter order information of another subband; and truncating the subband filter coefficient by using the obtained filter order information and a parameterization device therefor.
1. A method for generating a set of filter coefficients for an audio signal, comprising:
obtaining a set of subband filter coefficients for each subband, wherein the set of subband filter coefficients is obtained from a set of time domain binaural room impulse response (BRIR) filter coefficients;
obtaining average reverberation time information of each subband, wherein the average reverberation time information is an average value of reverberation time information extracted respectively from a set of subband filter coefficients for each channel in a corresponding subband;
determining a filter order of each subband based on average reverberation time information of a corresponding subband, wherein the filter order is determined to be variable in a frequency domain; and
truncating the set of subband filter coefficients by using the filter order of a corresponding subband.
2. The method of claim 1 , wherein the filter order is determined as a smaller value between a reference truncation length of the corresponding subband and an original length of the set of subband filter coefficients, and
wherein the reference truncation length of the corresponding subband is obtained by using the average reverberation time information.
3. The method of claim 2 , wherein when a length of the set of time domain BRIR filter coefficients is larger than a predetermined value, the reference truncation length is obtained by curve fitting the average reverberation time information.
4. The method of claim 3 , wherein the curve-fitted filter average reverberation time information is a value of power of 2 having an approximated integer value as an index, and wherein the approximated integer value is obtained by performing a polynomial curve fitting to the average reverberation time information.
5. The method of claim 2 , wherein when a length of the set of time domain BRIR filter coefficients is not larger than a predetermined value, the reference truncation length is obtained by using the average reverberation time information without performing a curve fitting.
6. The method of claim 5 , wherein the reference truncation length is determined to be a value of power of 2 having a log-scaled approximated integer value of the average reverberation time information as an index.
7. The method of claim 2 , wherein the reference truncation length is a value of power of 2.
8. An audio processing apparatus for generating a set of filter coefficients for an audio signal, the apparatus configured to:
obtain a set of subband filter coefficients for each subband wherein the set of subband filter coefficients is obtained from a set of time domain binaural room impulse response (BRIR) filter coefficients;
obtain average reverberation time information of each subband, wherein the average reverberation time information is an average vale of reverberation time information extracted respectively from a set of subband filter coefficients for each channel in a corresponding subband;
determine a filter order of each subband based at least in part on the average reverberation time information of a corresponding subband, wherein the filter order is determined to he variable in a frequency domain; and
truncates the set of subband filter coefficients by using the filter order of a corresponding subband.
9. The apparatus of claim 8 , wherein the filter order is determined as a smaller value between a reference truncation length of the corresponding subband and an original length of the set of subband filter coefficients, and
wherein the reference truncation length of the corresponding subband is obtained by using the average reverberation time information.
10. The apparatus of claim 9 , wherein when a length of the set of time domain BRIR filter coefficients is larger than a predetermined value, the reference truncation length is obtained by curve fitting the average reverberation time information.
11. The apparatus of claim 10 , wherein the curve-fitted average reverberation time information is a value of power of 2 having an approximated integer value as an index, and
wherein the approximated integer value is obtained by performing a polynomial curve-fitting to the average reverberation time information.
12. The apparatus of claim 9 , wherein when a length of the set of time domain BRIR filter coefficients is not larger than a predetermined value, the reference truncation length is obtained by using the average reverberation time information without performing a curve fitting.
13. The apparatus of claim 12 , wherein the reference truncation length is determined to be a value of power of 2 having a log-scaled approximated integer value of the average reverberation time information as an index.
14. The apparatus of claim 9 , wherein the reference truncation length is a value of power of 2.