Adaptive Microphone-Loudspeaker Arrays for Psychoacoustic Room Correction
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Solution Overview
Problem
Current room correction and equalization techniques face challenges in small enclosures like car cabins due to increased computational complexity and inefficiencies in mapping loudspeakers to listening positions, often excluding influential speakers beyond a given distance, and fail to utilize psychoacoustic principles for improved sound perception.
Innovation Solution
An audio system utilizing a plurality of loudspeakers and microphones, with an audio controller that determines psychoacoustic perceived loudness to map loudspeakers to microphones based on human perception, employing adaptive processes for equalization, including both fixed and adaptive filter coefficients to account for changing room conditions.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Ease of operation
If proximity analysis is used to map loudspeakers to listening positions, then the mapping process is simple, but influential speakers beyond a given distance are excluded and computational complexity increases in small enclosures
Solution Approach 1:
The patent changes the mapping parameter from physical distance to psychoacoustic perceived loudness (PPL). Instead of using proximity-based distance metrics, the system calculates PPL values that reflect how humans actually perceive sound intensity, incorporating frequency-weighting and temporal integration. This allows speakers beyond physical proximity thresholds to be included when they contribute significantly to the perceived sound field.
Solution Approach 2:
The patent replaces the mechanical/geometric proximity analysis system with a psychoacoustic measurement system. Rather than relying on physical distance calculations, the system uses PPL metrics that model human auditory perception, including frequency-dependent sensitivity and temporal integration effects. This substitution enables more accurate identification of influential speakers in complex acoustic environments.
2Reliability
If more features are supported in room correction systems, then sound reproduction quality improves, but computational complexity increases exponentially
Solution Approach 1:
The patent segments the room correction problem into distinct functional modules: PPL calculation for each speaker-microphone pair, speaker grouping based on PPL thresholds, and separate equalization processing for each group. This modular segmentation allows the system to handle multiple features (multi-position support, adaptive processing) while maintaining manageable computational complexity through divide-and-conquer strategies.
Solution Approach 2:
The patent applies partial action by selectively processing only the most influential speaker-microphone pairs based on PPL thresholds. Rather than attempting to model all possible acoustic paths and reflections, the system focuses computational resources on the dominant sound paths that contribute most significantly to the perceived audio quality, achieving high effectiveness with reduced complexity.
3Device complexity
If fixed room equalizers are used, then system complexity is reduced, but adaptability to changing room conditions deteriorates
Solution Approach 1:
The patent implements dynamic adaptability through continuous or periodic PPL measurements and equalization updates. The system can transition between fixed and adaptive modes, adjusting equalization parameters in response to changing room conditions such as temperature, humidity, furniture arrangement, or occupancy. This dynamic approach maintains low baseline complexity while enabling adaptability when needed.
Solution Approach 2:
The patent incorporates feedback mechanisms where microphones continuously monitor the acoustic environment and PPL values are recalculated based on measured room response. This feedback loop enables the system to detect changes in room conditions and automatically adjust equalization parameters, providing adaptability without requiring complex manual reconfiguration or excessive computational resources.
Data Source
AI summary
In at least one embodiment, an audio system is provided. The audio system includes a plurality of loudspeaker, a plurality of microphones, and an audio controller. The plurality of loudspeakers transmits an audio signal in a listening environment. The plurality of microphones detects the audio signal in the listening environment. The at least one audio controller is configured to determine a first psychoacoustic perceived loudness (PPL) of the audio signal as the audio signal is played back through a first loudspeaker of the plurality of loudspeakers and to determine a second PPL of the audio signal as the audio signal is sensed by a first microphone of the plurality of microphones. The at least one audio controller is further configured to map the first loudspeaker of the plurality of loudspeakers to the first microphone of the plurality of microphones based at least on the first PPL and the second PPL.


