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

• US 10,580,417 B2
• Filed: 10/22/2014
• Issued: 03/03/2020
• Est. Priority Date: 10/22/2013
• Status: Active Grant

• Current Assignee: Electronics and Telecommunications Research Institute, Industry Academic Cooperation Foundation, WILUS Inc.
• Original Assignee: Industry Academic Cooperation Foundation
• Inventors: Lee, Taegyu, Oh, Hyunoh, Park, Youngcheol, Youn, Daehee, Seo, Jeongil, Lee, Yongju, Beack, Seungkwon, Kang, Kyeongok, Jang, Daeyoung
• Primary Examiner(s): Nguyen, Duc
• Assistant Examiner(s): Eljaiek, Alexander L
• Application Number:
  US15/031,275
• Publication Number:
  US 20160277865A1
• Time in Patent Office: 1,959 Days
• Field of Search: 381 17, 381310, 381300, 381302, 381 1, 381 18, 381 22, 381303, 381307, 381309, 381 59, 700 94
• CPC Class Codes: G10H 2250/111 Impulse response, i.e. filters defined or specified by their temporal impulse response features, e.g. for echo or reverberation applications
• CPC Class Codes: G10H 2250/145 Convolution, e.g. of a music input signal with a desired impulse response to compute an output
• CPC Class Codes: G10L 19/008 Multichannel audio signal coding or decoding using interchannel correlation to reduce redundancy, e.g. joint-stereo, intensity-coding or matrixing
• CPC Class Codes: H04R 3/00 Circuits for transducers , loudspeakers or microphones
• CPC Class Codes: H04R 5/033 Headphones for stereophonic communication details thereof, e.g. relating to batteries, cables or control elements H04R1/10
• CPC Class Codes: H04S 2400/01 Multi-channel, i.e. more than two input channels, sound reproduction with two speakers wherein the multi-channel information is substantially preserved
• CPC Class Codes: H04S 2420/01 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 Class Codes: H04S 2420/03 Application of parametric coding in stereophonic audio systems
• CPC Class Codes: H04S 3/00 Systems employing more than two channels, e.g. quadraphonic H04S5/00, H04S7/00 take precedence
• CPC Class Codes: H04S 3/002 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 Class Codes: H04S 3/004 For headphones
• CPC Class Codes: H04S 3/008 in which the audio signals are in digital form, i.e. employing more than two discrete digital channels data reduction aspects thereof based on psychoacoustics G10L19/02

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.

Claims

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.