Method for generating filter for audio signal and parameterizing device therefor

Patent metadata

Field	Value
Publication number	US-10692508-B2
Application number	US-201816224820-A
Country	US
Kind code	B2
Filing date	Dec 19, 2018
Priority date	Oct 22, 2013
Publication date	Jun 23, 2020
Grant date	Jun 23, 2020

Abstract

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.

First claim

What is claimed is:

1. A method for post-processing an audio signal by a binaural renderer, comprising:
receiving an input audio signal;
receiving one or more binaural room impulse response (BRIR) filter coefficients in a time domain corresponding to at least one position in a virtual reproduction space;
converting the BRIR filter coefficients into a plurality of sets of subband filter coefficients;
truncating each set of subband filter coefficients based on a filter order value for each subband obtained by at least partially using characteristic information extracted from each set of subband filter coefficients, wherein the filter order value is determined to be variable in a frequency domain;
generating fast Fourier transform (FFT) filter coefficients by fast Fourier transforming each set of truncated subband filter coefficients by a predetermined block size in a corresponding subband; and
performing block-wise fast convolution on each subband signal of the input audio signal by using the FFT filter coefficients corresponding thereto,
wherein the predetermined block size is determined to be a smaller value between a first value and a second value,
wherein the first value is obtained by multiplying a reference filter length of a corresponding set of truncated subband filter coefficients by 2, and
wherein the second value is a predetermined maximum FFT size.

2. The method of claim 1 , wherein the reference filter length represents any one of a true value and an approximate value of the filter order value in a form of power of 2.

3. The method of claim 1 , wherein when the reference filter length is N and the predetermined block size corresponding thereto is M, the M is a power of 2 value and 2N=kM (k is a natural number).

4. The method of claim 1 , wherein the characteristic information includes reverberation time information of the corresponding set of subband filter coefficients.

5. The method of claim 1 , wherein the filter order value has a single value for each subband.

6. The method of claim 1 , wherein the generating FFT filter coefficients further comprising:
partitioning each set of truncated subband filter coefficients by a half of the predetermined block size;
generating temporary filter coefficients of the predetermined block size by using the partitioned filter coefficients, a first half part of the temporary filter coefficients being constituted by the partitioned filter coefficients and a second half part of the temporary filter coefficients being constituted by zero-padded values; and
generating the FFT filter coefficients by fast Fourier transforming the temporary filter coefficients.

7. An apparatus for post-processing an audio signal by a binaural renderer, comprising:
a first processor configured to generate a filter for an audio signal; and
a second processor configured to receive an input audio signal and filter the input audio signal by using one or more parameters generated by the first processor;
wherein the first processor is configured to:
receive one or more binaural room impulse response (BRIR) filter coefficients in a time domain corresponding to at least one position in a virtual reproduction space,
convert the BRIR filter coefficients into a plurality of sets of subband filter coefficients,
truncate each set of subband filter coefficients based on a filter order value for each subband obtained by at least partially using characteristic information extracted from each set of subband filter coefficients, wherein the filter order value is determined to be variable in a frequency domain, and
generate fast Fourier transform (FFT) filter coefficients by fast Fourier transforming each set of truncated subband filter coefficients by a predetermined block size in a corresponding subband,
wherein the second processor is configured to perform block-wise fast convolution on each subband signal of the input audio signal by using the FFT filter coefficients corresponding thereto,
wherein the predetermined block size is determined to be a smaller value between a first value and a second value,
wherein the first value is obtained by multiplying a reference filter length of a corresponding set of truncated subband filter coefficients by 2, and
wherein the second value is a predetermined maximum FFT size.

8. The apparatus of claim 7 , wherein the reference filter length represents any one of a true value and an approximate value of the filter order value in a form of power of 2.

9. The apparatus of claim 7 , wherein when the reference filter length is N and the predetermined block size corresponding thereto is M, the M is a power of 2 value and 2N=kM (k is a natural number).

10. The apparatus of claim 7 , wherein the characteristic information includes reverberation time information of the corresponding set of subband filter coefficients.

11. The apparatus of claim 7 , wherein the filter order value has a single value for each subband.

12. The apparatus of claim 7 , wherein the first processor is further configured to:
partition each set of truncated subband filter coefficients by a half of the predetermined block size,
generate temporary filter coefficients of the predetermined block size by using the partitioned filter coefficients, a first half part of the temporary filter coefficients being constituted by the partitioned filter coefficients and a second half part of the temporary filter coefficients being constituted by zero-padded values, and
generate the FFT filter coefficients by fast Fourier transforming the temporary filter coefficients.

Assignees

• Electronics & Telecommunications Res Inst
• Univ Yonsei Iacf
• Wilus Inst Standards & Tech Inc

Inventors

• Lee Taegyu
• Oh Hyunoh
• Seo Jeongil
• Lee Yongju
• Beack Seungkwon
• Kang Kyeongok
• Jang Daeyoung
• Park Youngcheol
• Youn Daehee

Classifications

• H04S3/00 Primary

  Systems employing more than two channels, e.g. quadraphonic (H04S5/00, H04S7/00 take precedence) · CPC title

• H04S3/004

  For headphones · CPC title

• H04S3/002

  Non-adaptive circuits, e.g. manually adjustable or static, for enhancing the sound image or the spatial distribution (control circuits for electronic adaptation of the sound field H04S7/30) · CPC title

• H04S2420/03

  Application of parametric coding in stereophonic audio systems · CPC title

• G10H2250/145

  Convolution, e.g. of a music input signal with a desired impulse response to compute an output · CPC title