Adaptive Filter Coefficient Control for Vehicle Audio Echo Cancellation

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Solution Overview

Problem

In vehicle-mounted voice recognition systems, echo cancellation using adaptive filters is challenging when the number of voice collection devices is less than the number of voice sources, making it difficult to identify and isolate target voice signals amidst surrounding noise.

Innovation Solution

An audio processing system with multiple microphones and adaptive filters that determines which audio components are dominant and adjusts filter coefficients accordingly, allowing for effective echo cancellation even with fewer microphones than voice sources.

Engineering Contradictions & Design Principles

VSEngineering Contradiction Analysis

1Measurement precision

If the number of voice collection devices is increased to match the number of voice sources, then the ability to identify and isolate target voice signals is improved, but the device complexity and cost increase

Engineering Contradiction:
Improvetarget voice identification accuracyVSAvoidnumber of microphones
Core Design Contradiction:
Measurement precisionVSDevice complexity

Solution Approach 1:

The patent changes the parameter of filter coefficients dynamically based on the determination result of audio component dominance. By adjusting the filter coefficients according to whether the first or second audio component is dominant, the system achieves accurate target voice isolation without needing to increase the number of microphones, thus resolving the contradiction between measurement precision and device complexity

Inventive Principle:
Principle #35Parameter changes

Solution Approach 2:

The system dynamically adapts its processing based on real-time analysis of audio component dominance. The filter coefficients are not fixed but are controlled based on the determination result, allowing the system to optimize target voice extraction adaptively with a limited number of microphones

Inventive Principle:
Principle #15Dynamics

2Reliability

If the number of reference signals is increased to match the number of voice sources, then the echo cancellation performance is improved, but the processing complexity and cost increase

Engineering Contradiction:
Improveecho cancellation performanceVSAvoidnumber of reference signals
Core Design Contradiction:
ReliabilityVSDevice complexity

Solution Approach 1:

The patent controls the filter coefficients based on the determination result of audio component dominance, allowing effective echo cancellation with a reduced number of reference signals. This parameter adaptation enables the system to achieve reliable echo cancellation without proportionally increasing the number of reference signals

Inventive Principle:
Principle #35Parameter changes

Solution Approach 2:

The system uses a partial approach by determining which audio component is dominant and adjusting processing accordingly, rather than processing all possible voice sources equally. This allows effective echo cancellation with fewer reference signals by focusing computational resources on the dominant audio component

Inventive Principle:
Principle #16Partial or excessive action

Data Source

PatentUS12125468B2Audio processing system, audio processing device, and audio processing method
Publication Date: 2024.10.22 PANASONIC AUTOMOTIVE SYST CO LTD
  • US12125468B2 patent drawing
  • US12125468B2 patent drawing
  • US12125468B2 patent drawing

AI summary

An audio processing system includes at least one first microphone, at least one adaptive filter, and a processor. The at least one first microphone acquires a first audio signal and outputs a first signal based on the first audio signal. The first audio signal includes at least one of a first audio component generated at a first position and a second audio component generated at a second position different from the first position. The first signal is input to the at least one adaptive filter. The at least one adaptive filter outputs a passing signal based on the first signal. The processor, when executing a program stored in a memory, performs: making a determination of which of the first audio component and the second audio component the first audio signal includes more; and controlling a filter coefficient of the adaptive filter based on a result of the determination.