Method for generating filter for audio signal, and parameterization device for same
First Claim
1. A method for processing an audio signal, comprising:
- receiving an input audio signal;
receiving binaural room impulse response (BRIR) filter coefficients;
obtaining flag information indicating whether a length of the BRIR filter coefficients is more than a predetermined value in a time domain;
converting the BRIR filter coefficients into a set of subband filter coefficients for each subband;
obtaining average reverberation time information of a subband by using reverberation time information extracted from the set of subband filter coefficients;
obtaining one or more coefficients for curve fitting the average reverberation time information;
obtaining filter order information for determining a truncation length of the set of subband filter coefficients, wherein the filter order information is obtained, based on the flag information, by using the average reverberation time information or by using the one or more coefficients, and the filter order is determined to be variable in a frequency domain;
truncating the set of subband filter coefficients by using the filter order information, wherein an energy compensation is performed to the truncated set of subband filter coefficients based on the flag information; and
filtering each subband signal of the input audio signal by using the truncated set of subband filter coefficients corresponding thereto.
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Abstract
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.
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Citations
20 Claims
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1. A method for processing an audio signal, comprising:
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receiving an input audio signal; receiving binaural room impulse response (BRIR) filter coefficients; obtaining flag information indicating whether a length of the BRIR filter coefficients is more than a predetermined value in a time domain; converting the BRIR filter coefficients into a set of subband filter coefficients for each subband; obtaining average reverberation time information of a subband by using reverberation time information extracted from the set of subband filter coefficients; obtaining one or more coefficients for curve fitting the average reverberation time information; obtaining filter order information for determining a truncation length of the set of subband filter coefficients, wherein the filter order information is obtained, based on the flag information, by using the average reverberation time information or by using the one or more coefficients, and the filter order is determined to be variable in a frequency domain; truncating the set of subband filter coefficients by using the filter order information, wherein an energy compensation is performed to the truncated set of subband filter coefficients based on the flag information; and filtering each subband signal of the input audio signal by using the truncated set of subband filter coefficients corresponding thereto. - View Dependent Claims (2, 3, 4, 5, 6, 7, 8, 9, 10)
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11. An apparatus for processing an audio signal, the apparatus configured to:
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receive an input audio signal; receive binaural room impulse response (BRIR) filter coefficients; obtain flag information indicating whether a length of the BRIR filter coefficients is more than a predetermined value in a time domain; convert the BRIR filter coefficients into a set of subband filter coefficients for each subband; obtain average reverberation time information of a subband by using reverberation time information extracted from the set of subband filter coefficients; obtain one or more coefficients for curve fitting the average reverberation time information; obtain filter order information for determining a truncation length of the set of subband filter coefficients, wherein the filter order information is obtained, based on the flag information, by using the average reverberation time information or by using the one or more coefficients, and the filter order is determined to be variable in a frequency domain; truncate the set of subband filter coefficients by using the filter order information, wherein an energy compensation is performed to the truncated set of subband filter coefficients based on the flag information; and filter each subband signal of the input audio signal by using the truncated set of subband filter coefficients corresponding thereto. - View Dependent Claims (12, 13, 14, 15, 16, 17, 18, 19, 20)
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Specification