Method and apparatus for binaural rendering audio signal using variable order filtering in frequency domain

Patent metadata

Field	Value
Publication number	US-10580417-B2
Application number	US-201415031275-A
Country	US
Kind code	B2
Filing date	Oct 22, 2014
Priority date	Oct 22, 2013
Publication date	Mar 3, 2020
Grant date	Mar 3, 2020

Abstract

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.

First claim

What is claimed is:

1. A method for binaural rendering an audio signal, comprising:
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 at least a portion of a corresponding set of subband filter coefficients obtained from binaural room impulse response (BRIR) filter coefficients for binaural filtering of the input audio signal, a length of the set of truncated subband filter coefficients being individually determined for each subband based on characteristic information extracted from the corresponding set of subband filter coefficients, the length of the set of truncated subband filter coefficients being determined to be variable in a frequency domain, and the set of truncated subband filter coefficients being constituted by one or more FFT filter coefficients generated by performing fast Fourier transform (FFT) by a predetermined block size in a corresponding subband;
generating at least one subframe for each subband by performing fast Fourier transform of each subband signal based on a predetermined subframe size;
generating at least one filtered subframe for each subband, each filtered subframe being generated by multiplying a corresponding subframe and the FFT filter coefficients;
inverse fast Fourier transforming the at least one filtered subframe for each subband; and
generating a filtered subband signal by overlap-adding the at least one inverse fast Fourier transformed subframe for each subband.

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

3. The method of claim 1 , wherein a length of a set of truncated subband filter coefficients of at least one subband is different from a length of a set of truncated subband filter coefficients of another subband.

4. The method of claim 1 , wherein the predetermined block size and the predetermined subframe size have values of power of 2.

5. The method of claim 1 , wherein the predetermined subframe size is determined based on the predetermined block size in the corresponding subband.

6. The method of claim 5 , the performing of the fast Fourier transform of each subband signal comprises:
partitioning each subband signal by the predetermined subframe size;
generating a temporary subframe including a first half part constituted by the partitioned subband signal and a second half part constituted by zero-padded values; and
fast Fourier transforming the generated temporary subframe.

7. An apparatus for processing an audio signal, which is used for performing binaural rendering of input audio signals, each input audio signal including a plurality of subband signals, the apparatus comprising:
a processor configured to perform rendering of a direct sound and early reflections sound parts for each subband signal, wherein the processor is further configured to:
receive an input audio signal;
receive 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 at least a portion of a corresponding set of subband filter coefficients obtained from binaural room impulse response (BRIR) filter coefficients for binaural filtering of the input audio signal, a length of the set of truncated subband filter coefficients being individually determined for each subband based on characteristic information extracted from the corresponding set of subband filter coefficients, the length of the set of truncated subband filter coefficients being determined to be variable in a frequency domain, and the set of truncated subband filter coefficients being constituted by one or more FFT filter coefficients generated by performing fast Fourier transform (FFT) by a predetermined block size in a corresponding subband;
generate at least one subframe for each subband by performing fast Fourier transform of each subband signal based on a predetermined subframe size;
generate at least one filtered subframe for each subband, each filtered subframe being generated by multiplying a corresponding subframe and the FFT filter coefficients;
inverse fast Fourier transform the at least one filtered subframe for each subband; and
generate a filtered subband signal by overlap-adding the at least one inverse fast Fourier transformed subframe for each subband.

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

9. The apparatus of claim 7 , wherein a length of a set of truncated subband filter coefficients of at least one subband is different from a length of a set of truncated subband filter coefficients of another subband.

10. The apparatus of claim 7 , wherein the predetermined block size and the predetermined subframe size have values of power of 2.

11. The apparatus of claim 7 , wherein the predetermined subframe size is determined based on the predetermined block size in the corresponding subband.

12. The apparatus of claim 11 , the performing of the fast Fourier transform of each subband signal comprises:
partitioning each subband signal by the predetermined subframe size;
generating a temporary subframe including a first half part constituted by the partitioned subband signal and a second half part constituted by zero-padded values; and
fast Fourier transforming the generated temporary subframe.

Assignees

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

Inventors

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

Classifications

• H04S2420/01

  Enhancing the perception of the sound image or of the spatial distribution using head related transfer functions [HRTF's] or equivalents thereof, e.g. interaural time difference [ITD] or interaural level difference [ILD] · CPC title

• H04R5/033

  Headphones for stereophonic communication {(details thereof, e.g. relating to batteries, cables or control elements H04R1/10)} · CPC title

• H04R3/00

  Circuits for transducers (arrangements for producing a reverberation or echo sound G10K15/08; amplifiers H03F) · CPC title

• 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