Adaptive Echo Filter With Frequency-Dependent Delay Handling

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Solution Overview

Problem

Existing echo canceller technologies in mobile audio devices are inadequate in effectively addressing the issue of acoustic coupling and echo suppression, particularly in closed rooms where long echoes are prevalent, leading to decreased audio quality.

Innovation Solution

An adaptive filter unit is designed with a processor that performs FFT transformations on audio signals, uses circular delay lines with varying lengths to handle frequency-dependent time delays, and subtracts delayed audio estimation data from transformed signals to filter out echoes, enhancing echo cancellation capabilities.

Engineering Contradictions & Design Principles

VSEngineering Contradiction Analysis

1Reliability

If conventional echo canceller applications are used in mobile audio devices, then basic echo suppression is provided, but audio quality deteriorates in closed rooms with long echoes

Engineering Contradiction:
Improveecho suppression effectivenessVSAvoidaudio quality degradation
Core Design Contradiction:
ReliabilityVSObject-affected harmful factors

Solution Approach 1:

The echo canceller is divided into multiple functional units: a filter unit for basic echo cancellation, a signal-to-echo ratio calculating unit for analyzing echo characteristics, and a residual echo suppressing unit for targeted suppression. This segmentation allows each unit to specialize in specific aspects of echo handling, improving overall effectiveness in complex acoustic environments like closed rooms.

Inventive Principle:
Principle #1Segmentation

Solution Approach 2:

The system continuously calculates the signal-to-echo ratio and uses this feedback to dynamically adjust the residual echo suppression. The feedback loop monitors the effectiveness of echo cancellation in real-time and modifies suppression parameters accordingly, maintaining high audio quality while effectively suppressing long echoes in closed rooms.

Inventive Principle:
Principle #23Feedback

2Object-generated harmful factors

If residual echo suppression is applied, then echo intensity is reduced, but signal processing complexity increases

Engineering Contradiction:
Improveecho intensityVSAvoidsignal processing complexity
Core Design Contradiction:
Object-generated harmful factorsVSDevice complexity

Solution Approach 1:

Instead of applying full echo cancellation across all frequency ranges and signal conditions, the system calculates the signal-to-echo ratio and applies residual echo suppression selectively only when and where echoes are actually present. This partial action approach reduces unnecessary processing complexity while maintaining effective echo intensity reduction.

Inventive Principle:
Principle #16Partial or excessive action

3Measurement precision

If signal-to-echo ratio calculation is performed, then echo characteristics are accurately identified, but processing time increases

Engineering Contradiction:
Improveecho detection accuracyVSAvoidprocessing time
Core Design Contradiction:
Measurement precisionVSLoss of time

Solution Approach 1:

The system performs preliminary FFT transformation and signal-to-echo ratio calculation on buffered audio data rather than processing individual samples in real-time. This preliminary action on pre-collected data maintains high echo detection accuracy while reducing the time pressure of real-time processing, allowing for more computationally intensive analysis.

Inventive Principle:
Principle #10Preliminary action

Data Source

PatentEP3273608B1An adaptive filter unit for being used as an echo canceller
Publication Date: 2022.01.12 SENNHEISER ELECTRONICS GMBH & CO KG
  • EP3273608B1 patent drawingFigure 1
  • EP3273608B1 patent drawingFigure 2
  • EP3273608B1 patent drawingFigure 3

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.