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1. 发明授权

US11195537B2 Method and apparatus for binaural rendering audio signal using variable order filtering in frequency domain 有权
公开(公告)号：US11195537B2
公开(公告)日：2021-12-07
申请号：US16747533
申请日：2020-01-21
申请人： INDUSTRY-ACADEMIC COOPERATION FOUNDATION, YONSEI UNIVERSITY , WILUS INSTITUTE OF STANDARDS AND TECHNOLOGY INC. , ELECTRONICS AND TELECOMMUNICATIONS RESEARCH INSTITUTE
发明人： Taegyu Lee , Hyunoh Oh , Youngcheol Park , Daehee Youn , Jeongil Seo , Yongju Lee , Seungkwon Beack , Kyeongok Kang , Daeyoung Jang
IPC分类号： G10L19/008 , H04S3/00 , H04R5/033 , H04R3/00
摘要： The present invention relates to a method and an apparatus for binaural rendering an audio signal using variable order filtering in frequency domain. To this end, provided are a method for processing an audio signal including: receiving an input audio signal; receiving a set of truncated subband filter coefficients for filtering each subband signal of the input audio signal, the set of truncated subband filter coefficients being constituted by one or more FFT filter coefficients generated by performing FFT by a predetermined block size; generating at least one subframe for each subband; generating at least one filtered subframe for each subband; performing inverse FFT on the filtered subframe for each subband; and generating a filtered subband signal by overlap-adding the transformed subframe for each subband and an apparatus for processing an audio signal using the same.

2. 发明授权

US10580417B2 Method and apparatus for binaural rendering audio signal using variable order filtering in frequency domain 有权
公开(公告)号：US10580417B2
公开(公告)日：2020-03-03
申请号：US15031275
申请日：2014-10-22
申请人： INDUSTRY-ACADEMIC COOPERATION FOUNDATION, YONSEI UNIVERSITY , WILUS INSTITUTE OF STANDARDS AND TECHNOLOGY INC. , ELECTRONICS AND TELECOMMUNICATIONS RESEARCH INSTITUTE
发明人： Taegyu Lee , Hyunoh Oh , Youngcheol Park , Daehee Youn , Jeongil Seo , Yongju Lee , Seungkwon Beack , Kyeongok Kang , Daeyoung Jang
IPC分类号： G10L19/008 , H04S3/00 , H04R5/033 , H04R3/00
摘要： The present invention relates to a method and an apparatus for binaural rendering an audio signal using variable order filtering in frequency domain. To this end, provided are a method for processing an audio signal including: receiving an input audio signal; receiving a set of truncated subband filter coefficients for filtering each subband signal of the input audio signal, the set of truncated subband filter coefficients being constituted by one or more FFT filter coefficients generated by performing FFT by a predetermined block size; generating at least one subframe for each subband; generating at least one filtered subframe for each subband; performing inverse FFT on the filtered subframe for each subband; and generating a filtered subband signal by overlap-adding the transformed subframe for each subband and an apparatus for processing an audio signal using the same.

3. 发明授权

US10692508B2 Method for generating filter for audio signal and parameterizing device therefor 有权
公开(公告)号：US10692508B2
公开(公告)日：2020-06-23
申请号：US16224820
申请日：2018-12-19
申请人： ELECTRONICS AND TELECOMMUNICATIONS RESEARCH INSTITUTE , INDUSTRY-ACADEMIC COOPERATION FOUNDATION, YONSEI UNIVERSITY , WILUS INSTITUTE OF STANDARDS AND TECHNOLOGY INC.
发明人： Taegyu Lee , Hyunoh Oh , Jeongil Seo , Yongju Lee , Seungkwon Beack , Kyeongok Kang , Daeyoung Jang , Youngcheol Park , Daehee Youn
IPC分类号： G06F17/00 , G10L19/008 , H04S3/00 , H04R5/033 , H04R3/00
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.

4. 发明授权

US10204630B2 Method for generating filter for audio signal and parameterizing device therefor 有权
公开(公告)号：US10204630B2
公开(公告)日：2019-02-12
申请号：US15031274
申请日：2014-10-22
申请人： ELECTRONICS AND TELECOMMUNICATIONS RESEARCH INSTITUTE , INDUSTRY-ACADEMIC COOPERATION FOUNDATION, YONSEI UNIVERSITY , WILUS INSTITUTE OF STANDARDS AND TECHNOLOGY INC.
发明人： Taegyu Lee , Hyunoh Oh , Jeongil Seo , Yongju Lee , Seungkwon Beack , Kyeongok Kang , Daeyoung Jang , Youngcheol Park , Daehee Youn
IPC分类号： G06F17/00 , G10L19/008 , H04S3/00 , H04R5/033 , H04R3/00
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.

5. 发明授权

US09860668B2 Audio signal processing method and device 有权
公开(公告)号：US09860668B2
公开(公告)日：2018-01-02
申请号：US15300277
申请日：2015-04-02
申请人： WILUS INSTITUTE OF STANDARDS AND TECHNOLOGY INC. , ELECTRONICS AND TELECOMMUNICATIONS RESEARCH INSTITUTE
发明人： Hyun Oh Oh , Taegyu Lee , Jeongil Seo
IPC分类号： H04S3/00 , H04S7/00
CPC分类号： H04S7/307 , G10L19/008 , H04R3/04 , H04R2430/03 , H04R2499/11 , H04R2499/15 , H04S3/008 , H04S7/303 , H04S7/306 , H04S2400/01 , H04S2400/03 , H04S2400/11 , H04S2420/01 , H04S2420/07
摘要： The present invention relates to a method and an apparatus for processing an audio signal, and more particularly, to a method and an apparatus for processing an audio signal, which synthesizes an object signal and a channel signal and effectively binaural-render the synthesized signal.To this end, the present invention provides a method for processing an audio signal, including: receiving an input audio signal including a multi-channel signal; receiving filter order information variably determined for each subband of a frequency domain; receiving block length information for each subband based on a fast Fourier transform length for each subband of filter coefficients for binaural filtering of the input audio signal; receiving Variable Order Filtering in Frequency-domain (VOFF) coefficients corresponding to each subband and each channel of the input audio signal per block of the corresponding subband, a total sum of lengths of the VOFF coefficients corresponding to the same subband and the same channel being determined based on the filter order information of the corresponding subband; and filtering each subband signal of the input audio signal by using the received VOFF coefficients to generate a binaural output signal and an apparatus for processing an audio signal by using the same.

6. 发明授权

US11343630B2 Audio signal processing method and apparatus 有权
公开(公告)号：US11343630B2
公开(公告)日：2022-05-24
申请号：US17197047
申请日：2021-03-10
申请人： WILUS INSTITUTE OF STANDARDS AND TECHNOLOGY INC.
发明人： Hyunoh Oh , Taegyu Lee , Jinsam Kwak , Juhyung Son
IPC分类号： H04S3/00 , G10L19/20
摘要： The present invention relates to a method and an apparatus for processing an audio signal, and more particularly, to a method and an apparatus for processing an audio signal, which synthesize an object signal and a channel signal and effectively perform binaural rendering of the synthesized signal.
To this end, provided are a method for processing an audio signal, which includes: receiving an input audio signal including a multi-channel signal; receiving truncated subband filter coefficients for filtering the input audio signal, the truncated subband filter coefficients being at least some of subband filter coefficients obtained from binaural room impulse response (BRIR) filter coefficients for binaural filtering of the input audio signal and the length of the truncated subband filter coefficients being determined based on filter order information obtained by at least partially using reverberation time information extracted from the corresponding subband filter coefficients; obtaining vector information indicating the BRIR filter coefficients corresponding to each channel of the input audio signal; and filtering each subband signal of the multi-channel signal by using the truncated subband filter coefficients corresponding to the relevant channel and subband based on the vector information and an apparatus for processing an audio signal by using the same.

7. 发明授权

US11109180B2 Method for generating filter for audio signal, and parameterization device for same 有权
公开(公告)号：US11109180B2
公开(公告)日：2021-08-31
申请号：US16864127
申请日：2020-04-30
申请人： WILUS INSTITUTE OF STANDARDS AND TECHNOLOGY INC.
发明人： Taegyu Lee , Hyunoh Oh
IPC分类号： H04S7/00 , G10L19/008 , H04S3/00 , G10L19/02
摘要： 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.

8. 发明授权

US10771910B2 Audio signal processing method and apparatus 有权
公开(公告)号：US10771910B2
公开(公告)日：2020-09-08
申请号：US16395242
申请日：2019-04-26
申请人： WILUS INSTITUTE OF STANDARDS AND TECHNOLOGY INC.
发明人： Hyunoh Oh , Taegyu Lee , Jinsam Kwak , Juhyung Son
IPC分类号： H04S3/00 , G10L19/20
摘要： The present invention relates to a method and an apparatus for processing an audio signal, and more particularly, to a method and an apparatus for processing an audio signal, which synthesize an object signal and a channel signal and effectively perform binaural rendering of the synthesized signal.To this end, provided are a method for processing an audio signal, which includes: receiving an input audio signal including a multi-channel signal; receiving truncated subband filter coefficients for filtering the input audio signal, the truncated subband filter coefficients being at least some of subband filter coefficients obtained from binaural room impulse response (BRIR) filter coefficients for binaural filtering of the input audio signal and the length of the truncated subband filter coefficients being determined based on filter order information obtained by at least partially using reverberation time information extracted from the corresponding subband filter coefficients; obtaining vector information indicating the BRIR filter coefficients corresponding to each channel of the input audio signal; and filtering each subband signal of the multi-channel signal by using the truncated subband filter coefficients corresponding to the relevant channel and subband based on the vector information and an apparatus for processing an audio signal by using the same.

9. 发明授权

US09848275B2 Audio signal processing method and device 有权
公开(公告)号：US09848275B2
公开(公告)日：2017-12-19
申请号：US15300273
申请日：2015-04-02
申请人： WILUS INSTITUTE OF STANDARDS AND TECHNOLOGY INC.
发明人： Taegyu Lee , Hyun Oh Oh
IPC分类号： H04S7/00 , H04S3/00
CPC分类号： H04S7/307 , G10L19/008 , H04R3/04 , H04R2430/03 , H04R2499/11 , H04R2499/15 , H04S3/008 , H04S7/303 , H04S7/306 , H04S2400/01 , H04S2400/03 , H04S2400/11 , H04S2420/01 , H04S2420/07
摘要： The present invention relates to a method and an apparatus for processing an audio signal, and more particularly, to a method and an apparatus for processing an audio signal, which synthesizes an object signal and a channel signal and effectively binaural-render the synthesized signal.To this end, the present invention provides a method for processing an audio signal, including: receiving an input audio signal including at least one of a multi-channel signal and a multi-object signal; receiving type information of a filter set for binaural filtering of the input audio signal, the type of the filter set being one of a finite impulse response (FIR) filter, a parameterized filter in a frequency domain, and a parameterized filter in a time domain; receiving filter information for binaural filtering based on the type information; and performing the binaural filtering for the input audio signal by using the received filter information, wherein when the type information indicates the parameterized filter in the frequency domain, in the receiving of the filter information, a subband filter coefficient having a length determined for each subband of a frequency domain is received, and in the performing of the binaural filtering, each subband signal of the input audio signal is filtered by using the subband filter coefficient corresponding thereto and an apparatus for processing an audio signal by using the same.

10. 发明授权

US10701511B2 Method for generating filter for audio signal, and parameterization device for same 有权
公开(公告)号：US10701511B2
公开(公告)日：2020-06-30
申请号：US16544832
申请日：2019-08-19
申请人： WILUS INSTITUTE OF STANDARDS AND TECHNOLOGY INC.
发明人： Taegyu Lee , Hyunoh Oh
IPC分类号： H04S7/00 , G10L19/008 , H04S3/00 , G10L19/02
摘要： 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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