Audio Signal Processing Apparatus for Clear Talker Utterance Capture

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Solution Overview

Problem

Existing audio signal processing technologies face challenges in accurately capturing the beginning of a talker's utterance while maintaining clarity, as current gain control methods either introduce time lag or reduce clarity due to voice leakage across microphones.

Innovation Solution

An audio signal processing apparatus that combines gating type and gain sharing type gain control by selecting a channel group based on volume levels and forming multiple sound collection beams, which narrows down the number of channels to improve clarity and capture the start of a talker's utterance effectively.

Engineering Contradictions & Design Principles

VSEngineering Contradiction Analysis

1Measurement precision

If gating type gain control is used to block non-talker audio signals, then clarity is improved, but time lag occurs from when a talker is changed to when the microphone gain increases

Engineering Contradiction:
ImproveclarityVSAvoidtime lag
Core Design Contradiction:
Measurement precisionVSLoss of time

Solution Approach 1:

The system performs preliminary gain adjustment before the talker switch is complete. By anticipating the talker change and pre-adjusting gains, the system eliminates the time lag that would otherwise occur when switching between talkers, while maintaining clarity through controlled gain distribution.

Inventive Principle:
Principle #10Preliminary action

Solution Approach 2:

The gain values are dynamically adjusted based on real-time talker detection and voice activity. Instead of static gating, the system continuously modifies gain parameters to smoothly transition between talkers, eliminating time lag while preserving clarity through adaptive control.

Inventive Principle:
Principle #15Dynamics

2Loss of time

If gain sharing type control is used to set gain according to each audio signal level, then time lag is reduced, but clarity is reduced when voice leaks across multiple microphones

Engineering Contradiction:
Improvetime lagVSAvoidclarity
Core Design Contradiction:
Loss of timeVSMeasurement precision

Solution Approach 1:

Different gain values are assigned to different microphone channels based on their specific characteristics and spatial relationships. The system identifies which microphones are capturing talker voice versus leakage, and applies localized gain control to each channel, maintaining overall responsiveness while preserving clarity by preventing leakage from dominating.

Inventive Principle:
Principle #3Local quality

Solution Approach 2:

The system dynamically changes gain parameters based on detected voice activity and talker identification. By adjusting gain values in response to real-time conditions, the system maintains responsiveness (reducing time lag) while preserving clarity through context-aware parameter modification.

Inventive Principle:
Principle #35Parameter changes

3Adaptability or versatility

If multiple microphones are used to capture talker voice, then coverage is improved, but clarity is reduced due to voice leakage across microphones

Engineering Contradiction:
ImprovecoverageVSAvoidclarity
Core Design Contradiction:
Adaptability or versatilityVSMeasurement precision

Solution Approach 1:

The audio signal processing is segmented by microphone channel, with each channel receiving individualized gain control based on its specific capture characteristics. This allows the system to utilize multiple microphones for broad coverage while maintaining clarity by processing and controlling each microphone's contribution separately.

Inventive Principle:
Principle #1Segmentation

Solution Approach 2:

The system applies gain control selectively to microphone channels, giving partial action to microphones that are capturing leakage versus those capturing primary talker voice. By using excessive gain on optimal microphones and suppressing others, the system achieves both broad coverage and high clarity.

Inventive Principle:
Principle #16Partial or excessive action

Data Source

PatentEP3644588B1Audio signal processing apparatus, mixer and audio signal processing method
Publication Date: 2024.05.01 YAMAHA CORP
  • EP3644588B1 patent drawingFigure 1
  • EP3644588B1 patent drawingFigure 2
  • EP3644588B1 patent drawingFigure 3A~3B

AI summary

An audio signal processing method includes selecting a channel group of at least two channels according to a predetermined reference, from among audio signals of at least three channels, and controlling a gain of the audio signal of each channel of selected channel group, according to a volume level of the audio signal of each channel of the channel group.