Loudspeaker Array Control Using Hybrid Filter Segmentation

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Solution Overview

Problem

Existing loudspeaker array technologies face high computational costs and reduced audio quality when attempting to adaptively control audio signals in real-time for multiple listeners with moving positions, due to the complexity of calculating accurate acoustic transfer functions and the need for multiple inverse filters.

Innovation Solution

The method involves using a hybrid approach with simple, frequency-independent filters for dependent filters and more complex, frequency-dependent filters for independent filters, allowing for improved audio quality without significantly increasing computational load by combining matrices C and G, where C represents a simplified model and G provides a more accurate representation of acoustic transfer functions.

Engineering Contradictions & Design Principles

VSEngineering Contradiction Analysis

1Measurement precision

If accurate acoustic transfer functions are used for real-time filter weight computation, then audio quality is improved, but computational cost increases significantly

Engineering Contradiction:
Improveaudio qualityVSAvoidcomputational cost
Core Design Contradiction:
Measurement precisionVSPower

Solution Approach 1:

The filter weight computation is segmented into two distinct stages: an offline pre-computation stage where accurate transfer functions are calculated and stored, and an online real-time stage where pre-computed weights are applied. This segmentation allows accurate audio reproduction without real-time computational burden.

Inventive Principle:
Principle #1Segmentation

Solution Approach 2:

Filter weights are pre-computed offline based on accurate acoustic transfer functions before actual audio reproduction. This preliminary action transfers the computational burden from real-time operation to an offline setup phase, enabling high-quality audio output with minimal real-time processing.

Inventive Principle:
Principle #10Preliminary action

2Adaptability or versatility

If filter weights are computed in real-time for moving control points, then adaptability is improved, but computational complexity increases

Engineering Contradiction:
Improvereal-time adaptabilityVSAvoidcomputational complexity
Core Design Contradiction:
Adaptability or versatilityVSDevice complexity

Solution Approach 1:

The system segments adaptability into two parts: pre-computed transfer functions that capture acoustic environment characteristics, and real-time filter weight application that adapts to listener positions. This allows adaptability without full real-time recomputation of transfer functions.

Inventive Principle:
Principle #1Segmentation

Solution Approach 2:

Pre-computed transfer functions serve as reusable templates that are applied to different listener positions through filter weight adjustments. Instead of computing new transfer functions for each position, the system copies and adapts the pre-computed models, reducing computational complexity.

Inventive Principle:
Principle #26Copying

3Measurement precision

If multiple inverse filters are used for cross-talk cancellation, then audio quality is improved, but processing complexity increases

Engineering Contradiction:
Improvecross-talk cancellation qualityVSAvoidprocessing complexity
Core Design Contradiction:
Measurement precisionVSDevice complexity

Solution Approach 1:

The cross-talk cancellation process is segmented into offline inverse filter computation and online application. The complex inverse filter calculations are performed once offline, while real-time operation simply applies these pre-computed filters, maintaining high cancellation quality with reduced processing complexity.

Inventive Principle:
Principle #1Segmentation

Data Source

PatentEP3920557B1Loudspeaker control
Publication Date: 2024.04.17 AUDIOSCENIC LTD
  • EP3920557B1 patent drawingFigure 1
  • EP3920557B1 patent drawingFigure 2
  • EP3920557B1 patent drawingFigure 3a~3b

AI summary

There is provided a method of controlling an array of loudspeakers. The method comprises: receiving a plurality of input audio signals to be reproduced, by the array, at a respective plurality of control points in an acoustic environment; and generating a respective output audio signal for each of the loudspeakers in the array by applying a set of filters to the plurality of input audio signals. The set of filters is based on: a first plurality of filter elements based on a first approximation of a set of transfer functions, each transfer function in the set of transfer functions being between an audio signal applied to a respective one of the loudspeakers and an audio signal received at a respective one of the control points from the respective one of the loudspeakers; and a second plurality of filter elements based on a second approximation of the set of transfer functions.