Binaural Rendering With Variable-Order Subband Filtering
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Solution Overview
Problem
Binaural rendering for multi-channel audio signals requires high computational complexity, especially with long binaural room impulse response filters, leading to inefficient processing and potential distortion, particularly in real-time environments.
Innovation Solution
The method involves using truncated subband filter coefficients derived from binaural room impulse response filters, which are processed through fast Fourier transform and inverse fast Fourier transform operations in subframes, allowing for efficient filtering with reduced computational complexity while maintaining sound quality.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Manufacturing precision
If long BRIR filter coefficients are used for binaural rendering, then sound quality is improved, but computational complexity increases
Solution Approach 1:
The patent divides the long BRIR filter into multiple subband filters operating in parallel across different frequency subbands. Each subband filter processes a specific frequency range, allowing the overall filtering operation to be distributed and computed more efficiently than a single long-time-domain filter, thus reducing computational complexity while maintaining sound quality.
Solution Approach 2:
The patent transforms the filtering operation from the time domain to the frequency domain using Fast Fourier Transform (FFT). By performing convolution in the frequency domain through point-wise multiplication, the computational complexity is reduced from O(N*M) for time-domain convolution to O(N log N + M log M) for FFT-based convolution, where N and M are the lengths of the signal and filter respectively.
2Device complexity
If truncated subband filter coefficients are used, then computational complexity decreases, but distortion increases
Solution Approach 1:
The patent applies different filter lengths to different subbands based on their specific requirements. Lower frequency subbands, which require longer filter lengths to capture reverberation characteristics, are processed with longer filters, while higher frequency subbands use shorter filters. This localized adaptation minimizes distortion in critical frequency regions while reducing overall computational complexity.
Solution Approach 2:
The patent dynamically adjusts the length of subband filter coefficients based on signal characteristics and processing requirements. By changing the filter length parameter adaptively rather than using a fixed uniform length across all subbands, the system optimizes the balance between computational efficiency and distortion minimization for different frequency regions.
Data Source
AI summary
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.