Method and device for audio signal processing

• US 9,961,469 B2
• Filed: 09/17/2014
• Issued: 05/01/2018
• Est. Priority Date: 09/17/2013
• Status: Active Grant

• Current Assignee: GCOA Company Limited, WILUS Inc.
• Original Assignee: WILUS Inc.
• Inventors: Lee, Taegyu, Oh, Hyunoh
• Primary Examiner(s): Ensey, Brian
• Assistant Examiner(s): Zhu, Qin
• Application Number:
  US15/022,922
• Publication Number:
  US 20160234620A1
• Time in Patent Office: 1,322 Days
• Field of Search: None
• 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: G10L 19/0204 using subband decomposition
• CPC Class Codes: G10L 25/48 specially adapted for particular use
• CPC Class Codes: H03H 17/0229 reducing the number of taps
• CPC Class Codes: H03H 17/0248 Filters characterised by a particular frequency response or filtering method
• CPC Class Codes: H03H 17/0266 Filter banks
• CPC Class Codes: H03H 17/0272 Quadrature mirror filters
• CPC Class Codes: H03H 21/0012 Digital adaptive filters
• CPC Class Codes: H04R 5/04 Circuit arrangements, e.g. for selective connection of amplifier inputs/outputs to loudspeakers, for loudspeaker detection, or for adaptation of settings to personal preferences or hearing impairments combinations of amplifiers H03F3/68; stereophonic systems H04S
• 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 2400/03 Aspects of down-mixing multi-channel audio to configurations with lower numbers of playback channels, e.g. 7.1 -> 5.1 H04S2400/01 takes precedence
• 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 2420/07 Synergistic effects of band splitting and sub-band processing
• CPC Class Codes: H04S 2420/11 Application of ambisonics 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 3/008 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
• CPC Class Codes: H04S 3/02 of the matrix type, i.e. in which input signals are combined algebraically, e.g. after having been phase shifted with respect to each other
• CPC Class Codes: H04S 5/005 of the pseudo five- or more-channel type, e.g. virtual surround
• CPC Class Codes: H04S 7/30 : Control circuits for electronic adaptation of the sound field

Abstract

The present invention relates to a method and an apparatus for processing a signal, which are used for effectively reproducing an audio signal, and more particularly, to a method and an apparatus for processing a signal, which are used for implementing binaural rendering for reproducing multi-channel or multi-object audio signals in stereo with a low calculation amount.

To this end, provided are a method for processing an audio signal including: receiving multi-audio signals including multi-channel or multi-object signals, each of the multi-audio signals including a plurality of subband signals, and the plurality of subband signals including a signal of a first subband group having low frequencies and a signal of a second subband group having high frequencies based on a predetermined frequency band; receiving at least one parameter corresponding to each subband signal of the second subband group, the at least one parameter being extracted from binaural room impulse response (BRIR) subband filter coefficients corresponding to each subband signal of the second subband group; and performing tap-delay line filtering of the subband signal of the second subband group by using the received parameter and an apparatus for processing an audio signal using the same.

Claims

1. A method for processing an audio signal, the method comprising:

receiving multi-audio signals including multi-channel or multi-object signals, each of the multi-audio signals including a plurality of subband signals, and the plurality of subband signals being classified into a first subband group having only low-frequency subband signals determined based on a predetermined frequency band and a second subband group having only high-frequency subband signals determined based on the predetermined frequency band;

performing, by a fast convolution unit, fast convolution on each low-frequency subband signal of the first subband group;

receiving, by a tap-delay line processing unit, at least one parameter corresponding to each high-frequency subband signal of the second subband group, the at least one parameter being extracted from binaural room impulse response (BRIR) subband filter coefficients corresponding to each high-frequency subband signal of the second subband group; and

performing, by the tap-delay line processing unit, one-tap-delay line filtering of each high-frequency subband signal of the second subband group by using the received parameter.

2. The method of claim 1, wherein the parameter includes one delay information for the corresponding BRIR subband filter coefficients and one gain information corresponding to the delay information.

3. The method of claim 2, wherein the delay information indicates positional information for a maximum peak in the BRIR subband filter coefficients.

4. The method of claim 2, wherein the delay information has a sample based integer value in a QMF domain.

5. The method of claim 2, wherein the gain information has a complex value.

6. The method of claim 1, further comprising:

summing the filtered high-frequency subband signals to 2-channel left and right subband signals for each subband;

coupling the summed left and right subband signals with left and right subband signals generated from the low-frequency subband signals of the first subband group; and

QMF-synthesizing the respective coupled left and right subband signals.

7. An apparatus for processing an audio signal, which is used for performing binaural rendering for multi-audio signals including multi-channel or multi-object signals, each of the multi-audio signals including a plurality of subband signals, and the plurality of subband signals including a first subband group having only low-frequency subband signals determined based on a predetermined frequency band and a second subband group having only high-frequency subband signals based on the predetermined frequency band, the apparatus comprising:

a fast convolution unit configured to perform fast convolution on each low-frequency subband signal of the first subband group; and

a tap-delay line processing unit configured to perform rendering of each high-frequency subband signal of the second subband group,

wherein the tap-delay line processing unit receives at least one parameter corresponding to each high-frequency subband signal of the second subband group, the at least one parameter being extracted from binaural room impulse response (BRIR) subband filter coefficients corresponding to each high-frequency subband signal of the second subband group, and

performs one-tap-delay line filtering of each high-frequency subband signal of the second subband group by using the received parameter.

8. The apparatus of claim 7, wherein the parameter includes one delay information for the corresponding BRIR subband filter coefficients and one gain information corresponding to the delay information.

9. The apparatus of claim 8, wherein the delay information indicates positional information for a maximum peak in the BRIR subband filter coefficients.

10. The apparatus of claim 8, wherein the delay information has a sample based integer value in a QMF domain.

11. The apparatus of claim 8, wherein the gain information has a complex value.

12. The apparatus of claim 7, wherein the tap-delay line processing unit sums the filtered high-frequency subband signals to 2-channel left and right subband signals for each subband, and

the apparatus for processing an audio signal further includes:

a mixer configured to couple the summed left and right subband signals with left and right subband signals generated from the low-frequency subband signals of the first subband group; and

a QMF synthesis unit configured to QMF-synthesize the respective coupled left and right subband signals.