Hearing Aid Sound Processing Using Time-Difference Distance Detection
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Solution Overview
Problem
Existing sound processing apparatuses face difficulties in effectively emphasizing sound from a speaker close to the user, especially when the microphones are closely spaced, as they require a large amplitude ratio and struggle in low frequency bands, making it challenging to distinguish between near and far sound sources.
Innovation Solution
The apparatus employs directional microphones and level calculation sections to determine the utterer's distance based on the difference between direct and reverberant sound levels, adjusting gain accordingly to emphasize the sound from the speaker close to the user, regardless of microphone distance.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Measurement precision
If microphones are disposed at a large distance to obtain a large amplitude ratio for near-field sound emphasis, then the near-field sound source separation performance is improved, but the device size increases and compactness is reduced
Solution Approach 1:
The patent changes the measurement parameter from amplitude ratio (which requires large microphone spacing) to time difference of arrival (which can be detected with small spacing). By measuring the time difference when sound waves arrive at each microphone and converting this to distance information, the system achieves near-field/far-field differentiation without requiring large physical separation between microphones, thus resolving the contradiction between measurement precision and device size
Solution Approach 2:
The patent replaces the mechanical approach of using large physical distance between microphones to achieve amplitude ratio differentiation with a temporal measurement approach. By substituting spatial separation with time-based measurement, the system can detect near-field sounds accurately while maintaining compact dimensions
2Volume of moving object
If microphones are disposed at a small distance to maintain compact device size, then the device compactness is improved, but the ability to distinguish between near and far sound sources deteriorates
Solution Approach 1:
The patent changes the detection parameter from amplitude (frequency-dependent and insensitive at small spacings) to time difference of arrival. The time-based measurement remains effective regardless of microphone spacing, allowing accurate distance discrimination in compact devices by measuring the temporal delay of sound wave arrival at each microphone
3Measurement precision
If amplitude ratio method is used for near-field sound emphasis, then the near-field sound can be emphasized, but the method becomes ineffective in low frequency bands where amplitude ratio is small
Solution Approach 1:
The patent substitutes the amplitude-based detection method with a time-based detection method. Since time difference of arrival is independent of frequency, this approach achieves frequency-independent near-field detection, resolving the limitation of amplitude ratio methods that fail in low frequency bands where amplitude differences are minimal
Applied Scientific Principles
This section explains which scientific principles are used to turn an abstract innovation direction into a practical engineering solution.
Function Achieved in This Case
This approach allows efficient emphasis of the sound from the conversational partner, effectively attenuating distant sounds, ensuring clear and accurate sound processing even with closely spaced microphones.
Implementation Method 1
a first directional microphone 101 for picking up a direct sound of a sound of an utterer; a second directional microphone 102 for picking up a reverberant sound of the sound of the utterer
Data Source
Figure 1
Figure 2(a)~2(b)
Figure 3(a)~3(b)
AI summary
The present invention provides a sound processing apparatus, a sound processing method and a hearing aid for efficiently emphasizing the sound of an utterer close to the user regardless of the distance between microphones. By using output signals from a plurality of omnidirectional microphones, respectively, the sound processing apparatus outputs a first directivity signal in which the main axis of directivity is formed in the direction of the utterer and outputs a second directivity signal in which the dead zone of directivity is formed in the direction of the utterer. The sound processing apparatus calculates the level of the first directivity signal and the level of the second directivity signal, and determines the distance to the utterer based on the level of the first directivity signal and the level of the second directivity signal. The sound processing apparatus derives a gain to be given to the first directivity signal according to the result of the determination and controls the level of the first directivity signal by using the gain.