Method and device for audio signal processing

• US 10,455,346 B2
• Filed: 04/02/2018
• Issued: 10/22/2019
• Est. Priority Date: 09/17/2013
• Status: Active Grant

• Current Assignee: GCOA Company Limited, Wilus Institute of Standards & Technology, Inc. (Good Day to Invent, Inc.)
• Original Assignee: Wilus Institute of Standards & Technology, Inc. (Good Day to Invent, Inc.)
• Inventors: Lee, Taegyu, Oh, Hyunoh
• Primary Examiner(s): Zhu, Qin
• Application Number:
  US15/942,588
• Publication Number:
  US 20180227692A1
• Time in Patent Office: 568 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 an input audio signal including a plurality of subband signals, wherein the plurality of subband signals include a first subband group containing only low-frequency subband signals determined based on a predetermined frequency band and a second subband group containing only high-frequency subband signals determined based on the predetermined frequency band;

performing a first filtering on each subband signal of the first subband group; and

performing a second filtering on each subband signal of the second subband group, wherein the second filtering is distinct from the first filtering,

wherein the first filtering is performed by using a first parameter set including one or more parameters for a corresponding subband and the second filtering is performed by using a second parameter set including one or more parameters for a corresponding subband,

wherein characteristics of the first parameter set is different from characteristics of the second parameter set,

wherein the second parameter set includes a set of gain and delay for the corresponding subband, and

wherein the second filtering comprises performing one-tap-delay line filtering on each subband signal of the second subband group by using the set of gain and delay for the corresponding subband.

2. The method of claim 1, wherein the number of one or more parameters of a second parameter set for a subband of the first subband group is equal to or less than the number of one or more parameters of a first parameter set for a subband of the first subband group.

3. The method of claim 1, wherein the first parameter set includes a set of subband filter coefficients obtained as at least a portion of proto-type subband filter coefficients for the corresponding subband, and

wherein the first filtering comprises performing fast convolution on each subband signal of the first subband group by using the set of subband filter coefficients for the corresponding subband.

4. The method of claim 3, wherein the proto-type subband filter coefficients are binaural room impulse response (BRIR) subband filter coefficients.

5. The method of claim 1, wherein the set of gain and delay is extracted from BRIR subband filter coefficients for the corresponding subband.

6. The method of claim 5, wherein the delay indicates positional information for a maximum peak in the BRIR subband filter coefficients.

7. The method of claim 1, wherein the gain has a complex value.

8. The method of claim 1, further comprising:

combining filtered subband signals of the first subband group and filtered subband signals of the second subband group into 2-channel left and right signals; and

performing a QMF-synthesis on the combined 2-channel left and right signals.

9. An apparatus for processing an audio signal, the apparatus comprising:

a binaural renderer configured to perform binaural rendering on an input audio signal,

wherein the binaural renderer is configured to;

receive an input audio signal including a plurality of subband signals, wherein the plurality of subband signals include a first subband group containing only low-frequency subband signals determined based on a predetermined frequency band and a second subband group containing only high-frequency subband signals determined based on the predetermined frequency band,

perform a first filtering on each subband signal of the first subband group, and

perform a second filtering on each subband signal of the second subband group, wherein the second filtering is distinct from the first filtering,

wherein the first filtering is performed by using a first parameter set including one or more parameters for a corresponding subband and the second filtering is performed by using a second parameter set including one or more parameters for a corresponding subband,

wherein characteristics of the first parameter set is different from characteristics of the second parameter set,

wherein the second parameter set includes a set of gain and delay for the corresponding subband, and

wherein the second filtering comprises performing one-tap-delay line filtering on each subband signal of the second subband group by using the set of gain and delay for the corresponding subband.

10. The apparatus of claim 9, wherein the number of one or more parameters of a second parameter set for a subband of the first subband group is equal to or less than the number of one or more parameters of a first parameter set for a subband of the first subband group.

11. The apparatus of claim 9, wherein the first parameter set includes a set of subband filter coefficients obtained as at least a portion of proto-type subband filter coefficients for the corresponding subband, and

wherein the first filtering comprises performing fast convolution on each subband signal of the first subband group by using the set of subband filter coefficients for the corresponding subband.

12. The apparatus of claim 11, wherein the proto-type subband filter coefficients are binaural room impulse response (BRIR) subband filter coefficients.

13. The apparatus of claim 9, wherein the set of gain and delay is extracted from BRIR subband filter coefficients for the corresponding subband.

14. The apparatus of claim 13, wherein the delay indicates positional information for a maximum peak in the BRIR subband filter coefficients.

15. The apparatus of claim 9, wherein the gain has a complex value.

16. The apparatus of claim 9, wherein the binaural renderer is further configured to:

combine filtered subband signals of the first subband group and filtered subband signals of the second subband group into 2-channel left and right signals, and

perform a QMF-synthesis on the combined 2-channel left and right signals.