Audio Precompensation Controller for Multi-Region Sound Field Control
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Solution Overview
Problem
Current audio precompensation technologies fail to effectively optimize sound field control in multichannel systems with a limited number of loudspeakers, particularly in car audio systems with awkward speaker positions and challenging acoustic environments, as existing methods lack control over time-domain properties and are sensitive to listener position.
Innovation Solution
A digital audio precompensation controller is designed using mathematical models and model-based optimization, estimating impulse responses at discrete measurement positions and adjusting filter parameters to optimize the sound field, allowing for unified control of equalization, crossover design, delay, and up-mixing, while ensuring stability and flexibility in loudspeaker placement.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Measurement precision
If traditional audio precompensation methods are used, then the system can operate with a limited number of loudspeakers, but the sound field control precision and time-domain properties cannot be effectively optimized
Solution Approach 1:
The patent segments the continuous sound field optimization problem into discrete regional optimizations. By dividing the listening space into multiple discrete regions and optimizing each region independently with separate filter sets, the system achieves precise sound field control in each zone without requiring excessive computational complexity for whole-space optimization.
Solution Approach 2:
The patent implements dynamic region activation based on detected listener positions. The system dynamically determines which listening regions are currently occupied and activates only the corresponding filter sets, rather than continuously optimizing all regions. This dynamic approach maintains high sound field control precision for active regions while reducing overall system complexity.
2Adaptability or versatility
If existing precompensation technologies are applied, then the system can handle awkward loudspeaker positions, but the methods lack control over time-domain properties and are sensitive to listener position
Solution Approach 1:
The patent pre-calculates and stores optimized filter sets for multiple discrete listening regions before operation. During actual use, the system simply selects the pre-computed filter set corresponding to the detected listener region, avoiding real-time optimization. This preliminary action ensures reliable sound field control across different listener positions while maintaining adaptability to various loudspeaker configurations.
Solution Approach 2:
The patent changes the fundamental parameter approach by discretizing the continuous listener position space into finite regions. Each region has its own pre-optimized filter parameters, transforming the problem from continuous optimization (which is sensitive to position variations) to discrete parameter selection (which is robust to position uncertainties).
3Measurement precision
If the system optimizes for multiple listening regions, then the sound field approximation improves across different positions, but the computational complexity and filter design difficulty increase
Solution Approach 1:
The patent divides the multi-region optimization problem into independent single-region optimizations. Each listening region is optimized separately with its own filter set, avoiding the computational burden of simultaneous multi-region optimization. The segmentation allows each sub-problem to be solved independently with standard optimization techniques.
Solution Approach 2:
The patent implements partial optimization by focusing computational resources on optimizing sound field accuracy only for regions that are actually occupied by listeners. Rather than continuously optimizing all possible regions, the system performs optimization only for active regions, reducing overall computational complexity while maintaining high accuracy where needed.
4Measurement precision
If continuous sound field optimization is attempted, then ideal sound field reconstruction is achieved, but the system requires ideal transducers and room acoustics that are never fulfilled in practice
Solution Approach 1:
The patent changes the fundamental assumption from continuous space optimization to discrete region optimization. By accepting that ideal continuous sound field reconstruction is unattainable with real-world transducers and room acoustics, the system parameters are changed to optimize for discrete, measurable regions where approximation is sufficient and practically achievable.
Solution Approach 2:
The patent applies partial optimization by targeting specific discrete listening regions rather than attempting complete continuous space optimization. This partial approach acknowledges practical limitations while achieving sufficient sound field reconstruction accuracy for the most important listening positions, making the system implementable with real-world constraints.
Data Source
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AI summary
The invention provides a scheme to design an audio precompensation controller for a multichannel audio system, with a prescribed number N of loudspeakers in prescribed positions so that listeners positioned in any of P>1 spatially extended listening regions should be given the illusion of being in another acoustic environments that has L sound sources (virtual loudspeakers) located at prescribed positions in a prescribed room acoustics. The invention provides a unified joint solution to the problems of equalizer design, crossover design, delay and level calibration, sum-response optimization and up-mixing. A multi-input multi-output audio precompensation controller is designed for an associated sound generating system comprising a limited number of loudspeaker inputs for emulating a number of virtual sound sources. The scheme for designing is based on: (S1) estimating, for each loudspeaker input signals, an impulse response at each of a set of measurement positions that cover the P listening regions; (S2) specifying a target impulse response (target stages) for each virtual sound source at each measurement position; and (S3) determining adjustable filter parameters of the audio precompensation controller so that a criterion function is optimized. The criterion function includes a weighted summation of powers of differences between the compensated estimated impulse responses and the target impulse responses over a discrete grid of said M measurement positions. This optimization can furthermore be iterated with a tuning of adjustable parameters in the target stages to improve the attainable criterion value and the accuracy of the compensation.