Adaptive Echo Filter With Frequency-Dependent Delay Allocation

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Solution Overview

Problem

Existing mobile audio devices face decreased audio quality due to acoustic coupling between speakers and microphones, particularly in closed rooms where echoes tend to be long, and current echo canceller technologies are not optimized for frequency-dependent time delays.

Innovation Solution

An adaptive filter unit with a processor that performs FFT transformations to calculate frequency-dependent time delays, allowing for efficient use of processing resources by allocating larger delays to low frequencies and smaller delays to higher frequencies, and integrating this unit into hearing devices for improved echo cancellation.

Engineering Contradictions & Design Principles

VSEngineering Contradiction Analysis

1Reliability

If frequency-dependent time delays are implemented in the adaptive filter unit, then echo cancellation performance is improved, but device complexity increases

Engineering Contradiction:
Improveecho cancellation performanceVSAvoiddevice complexity
Core Design Contradiction:
ReliabilityVSDevice complexity

Solution Approach 1:

The patent applies local quality by implementing different time delay values for different frequency bands. The adaptive filter unit processes low frequencies with longer time delays and high frequencies with shorter time delays, rather than applying a uniform time delay across all frequencies. This frequency-dependent approach optimizes echo cancellation for each frequency band according to its specific characteristics, improving overall performance while managing complexity through targeted differentiation.

Inventive Principle:
Principle #3Local quality

2Reliability

If larger time delays are allocated to low frequencies, then echo cancellation effectiveness is improved, but processing time increases

Engineering Contradiction:
Improveecho cancellation effectivenessVSAvoidprocessing time
Core Design Contradiction:
ReliabilityVSLoss of time

Solution Approach 1:

The patent changes the time delay parameter dynamically based on frequency content. Low frequency components are processed with larger time delays to effectively cancel long echoes characteristic of closed rooms, while high frequency components use smaller time delays to minimize processing time. This parameter adaptation allows the system to optimize between cancellation effectiveness and processing speed for different frequency ranges.

Inventive Principle:
Principle #35Parameter changes

3Manufacturing precision

If frequency-dependent time delay processing is implemented, then audio quality is enhanced, but computational resources are consumed

Engineering Contradiction:
Improveaudio qualityVSAvoidcomputational resources
Core Design Contradiction:
Manufacturing precisionVSUse of energy by moving object

Solution Approach 1:

The patent applies local quality by implementing different time delay values for different frequency bands. The adaptive filter unit processes low frequencies with longer time delays and high frequencies with shorter time delays, rather than applying a uniform time delay across all frequencies. This frequency-dependent approach optimizes echo cancellation for each frequency band according to its specific characteristics, improving overall performance while managing complexity through targeted differentiation.

Inventive Principle:
Principle #3Local quality

Data Source

PatentUS10506105B2Adaptive filter unit for being used as an echo canceller
Publication Date: 2019.12.10 SENNHEISER ELECTRONICS GMBH & CO KG
  • US10506105B2 patent drawing
  • US10506105B2 patent drawing
  • US10506105B2 patent drawing

AI summary

The invention relates to an adaptive filter unit, in particular for being used as an echo canceller, comprising a first filter input, configured to receive a first electric audio signal, indicative of a first audio signal A(t), a second filter input, configured to receive a second electric audio signal, indicative of a second audio signal B(t), a processor and a filter output. The processor is configured to calculate and provide audio estimation data X(fn, A(t1, . . . , tM(fn))) in the frequency domain; to calculate a transformed second audio signal Y(fn, B(t)), formed by a transformation of the second audio signal B(t) into the frequency domain; and to calculate a filtered audio signal by subtracting delayed audio estimation data from the transformed second audio signal, wherein the delayed audio estimation data is provided by a memory unit of the adaptive filter unit, which is arranged to provide a data exchange with the processor, and wherein the delayed audio estimation data comprises a frequency dependent time delay compared to the transformed second audio signal.