Room Audio Equalization Using Multiband Inverse Filtering
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Solution Overview
Problem
Audio signals are distorted when played in a room due to the room's transfer function and reflections, requiring an effective inverse filter to equalize the sound, but existing solutions face challenges like audible artifacts, high computational resources, and instability with long adaptive filters.
Innovation Solution
An audio enhancement system using analysis and synthesis filters with adaptive algorithms like FXLMS, poly-phase filters, and multi-rate systems to invert the unknown transfer function, achieving linearization while minimizing computational resources and signal delay.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Measurement precision
If an inverse filter is employed to equalize the coloring caused by the transfer function, then the aural characteristics are accurately reproduced, but the filter requires high computational resources and may produce audible artifacts
Solution Approach 1:
The patent divides the audio signal into multiple frequency bands using filter banks (analysis filters). Each band is processed independently by separate inverse filters, allowing parallel computation that reduces overall computational complexity while maintaining accurate reproduction of aural characteristics across the full frequency spectrum.
Solution Approach 2:
The patent employs adaptive filtering algorithms that dynamically adjust filter coefficients based on the actual acoustic environment. This dynamic adaptation allows the system to achieve accurate equalization with optimized computational resources, as the filters only compute what is necessary for the specific listening conditions rather than processing all frequencies at maximum detail.
2Measurement precision
If a long adaptive filter is used to compensate for room distortions, then the equalization accuracy is improved, but the filter becomes unstable and produces audible artifacts
Solution Approach 1:
Instead of using one long adaptive filter, the patent segments the frequency spectrum into multiple bands, each handled by a shorter adaptive filter. This segmentation maintains equalization accuracy across the full spectrum while improving stability, as each individual filter operates within a narrower frequency range where convergence is more reliable and less prone to producing audible artifacts.
Solution Approach 2:
The patent applies adaptive filtering selectively to specific frequency bands where room distortions are most problematic, rather than attempting to equalize all frequencies uniformly. This partial action approach achieves sufficient equalization accuracy for the most critical frequency ranges while avoiding the instability issues that arise from over-processing less problematic frequency ranges.
3Measurement precision
If existing inverse filtering methods are applied, then room distortions are compensated, but signal delay increases and computational resources are consumed
Solution Approach 1:
The patent processes different frequency bands in parallel through separate filter banks, allowing simultaneous computation rather than sequential processing. This parallel architecture significantly reduces the overall signal delay while maintaining accurate distortion compensation, as multiple frequency components are equalized concurrently rather than one after another.
Solution Approach 2:
The patent transforms the filtering operation into the frequency domain using Fourier transforms, where convolution operations become simple multiplications. This parameter transformation from time-domain to frequency-domain processing dramatically reduces computational complexity and signal delay while preserving the accuracy of distortion compensation.
Data Source
AI summary
An audio enhancement system is provided for compensating for distortions (e.g., linear distortions) of a sound signal reproduced by an audio system in a listening room. The audio enhancement system includes analysis filters that generate a plurality of analysis output signals from an audio signal to be enhanced. The system also includes synthesis filters that generate an enhanced audio signal from a number of synthesis input signals. The number of analysis output signals and the number of synthesis input signals preferably are equal. Signal processing elements between the analysis filters and the synthesis filters generate one of the synthesis input signals from a respective one of the analysis output signals to perform an inverse filtering for linearizing an unknown transfer function indicative of the audio system and the listening room in the respective frequency range.