Subband Audio Filter Parameterization for Low-Complexity Binaural Rendering

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Solution Overview

Problem

Binaural rendering for multi-channel audio signals requires high computational complexity, especially when using binaural room impulse response filters, which is inefficient and leads to significant computational overhead, particularly in real-time processing environments.

Innovation Solution

A method and apparatus for processing audio signals that utilize a binaural renderer with a BRIR parameterization unit to convert time-domain BRIR filters into QMF-domain filters, allowing for efficient binaural rendering by performing fast convolution and late reverberation generation in the QMF domain, reducing computational complexity while maintaining high sound quality.

Engineering Contradictions & Design Principles

VSEngineering Contradiction Analysis

1Reliability

If time-domain BRIR filtering is used for binaural rendering, then sound quality is maintained, but computational complexity becomes enormous

Engineering Contradiction:
Improvesound qualityVSAvoidcomputational complexity
Core Design Contradiction:
ReliabilityVSDevice complexity

Solution Approach 1:

The patent replaces time-domain convolution operations with frequency-domain filtering operations. Specifically, it substitutes the mechanical computation of long-time convolution (which requires enormous computational resources) with frequency-domain filter applications that achieve the same acoustic effect with much lower computational complexity. The BRIR filter is transformed into frequency domain through FFT, and the filtering operation is performed by simple multiplication in frequency domain rather than time-domain convolution.

Inventive Principle:
Principle #28Mechanics substitution (Replace mechanical system)

Solution Approach 2:

The patent changes the domain parameter from time domain to frequency domain. By applying FFT to transform the BRIR impulse response from time domain to frequency domain, the system changes the operational parameter space. This transformation allows the same filtering function to be achieved with different computational characteristics - specifically, replacing O(N*M) time-domain convolution complexity with O(N log N) frequency-domain processing complexity.

Inventive Principle:
Principle #35Parameter changes

2Productivity

If fast convolution based on FFT is used, then convolution efficiency is improved, but additional QMF synthesis operations are required increasing overall complexity

Engineering Contradiction:
Improveconvolution efficiencyVSAvoidoverall processing complexity
Core Design Contradiction:
ProductivityVSDevice complexity

Solution Approach 1:

The patent merges the binaural rendering filtering operation with the existing QMF synthesis operation that is already required by the audio codec. Instead of performing separate time-domain convolution and then separate QMF synthesis, the invention integrates both operations into a single frequency-domain filtering step. The BRIR frequency response is applied directly to the QMF subband signals, combining what would otherwise be two distinct processing stages into one unified operation.

Inventive Principle:
Principle #5Merging (Combining)

Solution Approach 2:

The patent makes the frequency-domain filter serve multiple functions simultaneously. The same frequency-domain filter coefficients are used both for implementing the binaural spatial rendering and for performing the QMF synthesis required by the audio decoding standard. This multi-functionality eliminates redundant operations and reduces overall system complexity while maintaining both spatial audio quality and decoding compatibility.

Inventive Principle:
Principle #6Universality (Multi-functionality)

Data Source

PatentEP3062534B1Method for generating filter for audio signal and parameterizing device therefor
Publication Date: 2021.03.03 WILUS INSTITUTE OF STANDARDS & TECHNOLOGY INC
  • EP3062534B1 patent drawingFigure 1
  • EP3062534B1 patent drawingFigure 2
  • EP3062534B1 patent drawingFigure 3

AI summary

The present invention relates to a method and an apparatus for processing a signal, which are used to effectively reproduce an audio signal, and more particularly, to a method for generating a filter for an audio signal, which are used for implementing a filtering for input audio signals with a low computational complexity and a parameterization apparatus therefor. 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.