MIMO Sound Field Generation with MELMS Algorithm
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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 impracticality of acoustic treatment in environments like vehicles, leading to distortions from room reflections.
Innovation Solution
A Multiple-Input Multiple-Output (MIMO) system employing a Multiple Error Least Mean Square (MELMS) algorithm with psychoacoustic constraints, including pre-ringing and magnitude constraints, to optimize loudspeaker signals and reduce the number of required loudspeakers while minimizing distortions from room reflections.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Manufacturing precision
If wave field synthesis or Ambisonics techniques are used to achieve high-quality spatial sound reproduction, then spatial reproduction quality is improved, but the number of loudspeakers required increases significantly
Solution Approach 1:
The patent transforms the sound reproduction problem from the spatial domain to the frequency domain by applying Fourier transforms. This parameter transformation allows the system to process and reproduce sound fields using fewer loudspeakers by manipulating frequency components rather than directly controlling spatial distribution across many speakers.
Solution Approach 2:
The patent replaces the mechanical approach of using many physical loudspeakers arranged in specific geometries with a signal processing approach. By using Fourier transforms and frequency domain manipulation, the system achieves spatial reproduction quality without requiring the physical presence of numerous loudspeakers in precise configurations.
2Manufacturing precision
If acoustic treatment is applied to eliminate wall reflections and achieve free-field conditions, then sound field accuracy is improved, but cost and practicality deteriorate in environments like vehicles
Solution Approach 1:
The patent replaces physical acoustic treatment (mechanical/structural modifications to the environment) with signal processing techniques. By transforming the problem into the frequency domain and using mathematical operations, the system compensates for room reflections and achieves accurate sound field reproduction without requiring expensive acoustic treatment of the physical space.
Solution Approach 2:
The patent introduces an intermediary signal processing layer (Fourier transforms and frequency domain manipulation) between the sound source and the listening environment. This intermediary processing allows the system to overcome adverse environmental conditions like wall reflections without physically modifying the environment, making the solution practical for vehicles and other untreated spaces.
3Ease of operation
If the listening room exhibits wall reflections, then implementation simplicity is improved, but sound field distortion increases
Solution Approach 1:
The patent replaces physical acoustic treatment with signal processing. By using Fourier transforms to move into the frequency domain, the system can mathematically compensate for distortions caused by wall reflections, maintaining implementation simplicity while eliminating sound field distortion through computational rather than physical means.
Solution Approach 2:
The patent employs frequency domain analysis and manipulation as a form of feedback control. By transforming the sound field information, analyzing it in the frequency domain, and applying appropriate transformations, the system continuously corrects for distortions introduced by room reflections, maintaining accuracy despite simple physical environments.
Data Source
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AI summary
A system and a method are configured to generate a sound wave field around a listening position in a room in which a loudspeaker array of K ≥ 1 groups of loudspeakers, with each group of loudspeakers having at least one loudspeaker, is disposed around a listening position, and a microphone array of M ≥ 1 groups of microphones, with each group of microphones having at least one microphone, is disposed at the listening position. The system and a method include equalizing filtering with controllable transfer functions in signal paths upstream of the K groups of loudspeakers and downstream of an input signal path, and controlling with equalization control signals of the controllable transfer functions for equalizing filtering 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. At least one of the K groups of loudspeakers is operated at a distance of less than 0.5 meters from the listening position.