Sound Collection Apparatus Using Directionality Formation and Delay Correction
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Solution Overview
Problem
Conventional sound collection methods using beam formers struggle to accurately collect target area sounds while suppressing non-target area sounds, especially in environments with strong reverberations, leading to potential distortions and incomplete suppression of non-target sounds.
Innovation Solution
A sound collection apparatus and method that form a directionality in the target area using multiple microphone arrays, correct delays, and apply an area sound enhancement filter to suppress non-target sounds by calculating a power ratio between the arrays' outputs, effectively reducing distortions and emphasizing target sounds.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Quantity of substance
If a subtraction-type beam former is used to form directionality, then the number of microphones required is reduced, but non-target area sounds in the same direction as the target area cannot be sufficiently suppressed
Solution Approach 1:
The patent divides the sound collection space into multiple regions (target area and non-target areas) and applies different processing to sounds from different regions. By segmenting the directional processing, the system can suppress non-target sounds in the same direction as target sounds while maintaining effective target sound collection with a limited number of microphones.
Solution Approach 2:
The patent transitions from conventional two-dimensional directional processing (azimuth only) to three-dimensional spatial processing by incorporating elevation angle information. This dimensional expansion enables the system to distinguish between target sounds and non-target sounds in the same azimuth direction by analyzing their different spatial origins.
2Measurement precision
If conventional spectral subtraction is applied twice for target area sound extraction, then target area sounds can be collected, but output target sounds become distorted
Solution Approach 1:
The patent introduces a feedback mechanism where the system estimates the spatial distribution of sounds based on microphone array inputs, uses this estimation to generate enhancement filters, and applies these filters to the original inputs. This closed-loop approach allows the system to extract target area sounds accurately while preserving their original characteristics through iterative refinement.
Solution Approach 2:
The patent changes the parameters of sound enhancement by dynamically adjusting filter characteristics based on estimated spatial sound distributions. Instead of applying fixed spectral subtraction, the system adapts filter parameters (such as gain and frequency response) according to the spatial location and intensity of sound sources, thereby extracting target sounds without distortion.
3Measurement precision
If directionalities from different directions are directed to a target area using multiple microphone arrays, then target area sounds can be collected, but non-target area sounds remain without sufficient suppression in strong reverberation environments
Solution Approach 1:
The patent performs preliminary estimation of the spatial distribution of sounds before applying enhancement filters. By predicting the locations and characteristics of target and non-target sound sources in advance, the system can pre-compute appropriate enhancement filters that suppress reverberation and non-target sounds while preserving target sounds, thereby improving performance in strong reverberation environments.
Solution Approach 2:
The patent introduces spatial distribution estimation as an intermediary process between microphone input and sound enhancement output. This intermediary estimation step analyzes the spatial characteristics of sounds and uses this information to guide the enhancement process, enabling effective suppression of non-target sounds and reverberation while maintaining target sound quality.
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
The solution effectively reduces distortions of target area sounds and suppresses non-target sounds even in environments with strong reverberations, improving the accuracy of sound collection by forming a filter based on the power ratio of beam former outputs.
Implementation Method 1
A BF is technology that forms a directionality by using a time difference of signals arriving at a plurality of microphones
Implementation Method 2
The time difference is calculated by the following Formula (1). τi=(d sin θL)/c (1) In Formula (1), d is a distance between the microphones M1 and M2, c is the speed of sound
Data Source
AI summary
There is provided a sound collection apparatus, including a directionality formation unit configured to form a directionality in a direction of a target area for input signals from a plurality of microphone arrays, a target area sound extraction unit configured to correct a delay between a target area and each of the microphone arrays, and a power of a target area sound component for an output from the directionality formation unit, suppress a non-target area sound by using each output after correction, and extract a target area sound, an area sound enhancement filter formation unit, and an area sound emphasis unit.


