Adaptive Sound Wave Field Generation Using MIMO Control
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Solution Overview
Problem
Current spatial sound field reproduction techniques, such as wave field synthesis and Ambisonics, face limitations in achieving high-quality spatial sound reproduction due to the need for a large number of loudspeakers and the distortion caused by room reflections, which is impractical in environments like cars without significant acoustic treatment.
Innovation Solution
A system and method using a loudspeaker array and a microphone array with adaptive equalizing filter modules to control sound wave propagation, employing a Multiple-Input Multiple-Output (MIMO) system with a Multiple Error Least Mean Square (MELMS) algorithm to optimize sound wave field generation around a listener's position, reducing the impact of room reflections and improving sound localization.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Measurement precision
If wave field synthesis or Ambisonics is used to achieve high-quality spatial sound reproduction, then sound field detail and spatial accuracy are improved, but the number of loudspeakers required increases significantly
Solution Approach 1:
The patent implements dynamic adaptation by using an adaptive control algorithm that continuously adjusts the transfer functions of equalizing filter modules based on real-time error signals from microphone arrays. This allows the system to optimize sound field reproduction dynamically without requiring a fixed large number of loudspeakers, resolving the contradiction between spatial accuracy and device complexity
Solution Approach 2:
The system employs feedback mechanisms where microphone arrays capture error signals from the reproduced sound field, which are then processed by adaptive control algorithms to adjust the loudspeaker signals. This closed-loop feedback enables high spatial reproduction accuracy with fewer loudspeakers by continuously optimizing the sound field based on actual measurements
2Measurement precision
If traditional equalization is used to compensate for room reflections, then sound field accuracy is improved, but the system becomes less adaptable to different acoustic environments
Solution Approach 1:
The patent transitions from static equalization to dynamic adaptive equalization where transfer functions are continuously adjusted based on real-time error signals from microphone arrays. This allows the system to adapt to different acoustic environments (such as car interiors vs. recording studios) while maintaining high sound field accuracy, resolving the contradiction between precision and adaptability
Solution Approach 2:
The system changes the parameters of equalizing filter modules dynamically based on the acoustic environment. The adaptive control algorithm modifies transfer function parameters in real-time according to the specific acoustic characteristics of the environment, enabling both high accuracy and environmental versatility
Applied Scientific Principles
This section explains which scientific principles are used to turn an abstract innovation direction into a practical engineering solution.
Function Achieved in This Case
The solution effectively generates a high-quality sound wave field with improved sound localization and reduced distortion, even in environments with limited acoustic treatment, by adaptively controlling the sound wave propagation and minimizing the effects of room reflections.
Implementation Method 1
generate a sound wave field around a listening position
Implementation Method 2
microphone array of M≥1 groups of microphones, with each group of microphones having at least one microphone, is disposed at the listening position
Implementation Method 3
K equalizing filter modules that are arranged in signal paths upstream of the groups of loudspeakers and downstream of an input signal path and that have controllable transfer functions
Implementation Method 4
K filter control modules that are arranged in signal paths downstream of the groups of microphones and downstream of the input signal path and that control the transfer functions of the K equalizing filter modules according to an adaptive control algorithm based on error signals from the K groups of microphones and an input signal on the input signal path
Data Source
AI summary
A system and method is configured to generate a sound wave field around a listening position in a target loudspeaker-room-microphone system in which a loudspeaker array of K≥1 groups of loudspeakers is disposed around the listening position, and a microphone array of M≥1 groups of microphones is disposed at the listening position. The system and method include equalizing filtering with controllable transfer functions in signal paths upstream of the K groups of loudspeakers. The system and method further include controlling with equalization control signals of the controllable transfer functions for equalizing filtering according to an adaptive control algorithm based on error signals and an input signal. The microphone array includes at least two first groups of microphones that are annularly disposed around a listener's head, around or in an artificial head or around or in a rigid sphere.