Adaptive Audio Filtering for Low-Latency Target Sound Control
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Solution Overview
Problem
Existing sound control technologies, such as those using equalizers and neural networks, face challenges in efficiently controlling audio signals in real-time applications like games and voice calls without increasing hardware scale and costs, and often result in delays that affect the real-time property of these applications.
Innovation Solution
An information processing apparatus that includes a filter processing unit and a filter setting unit, which uses an estimation algorithm to determine filter settings for controlling target audio sources within input signals, allowing for real-time sound control without the need for additional hardware like microphones and minimizing delays.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Object-affected harmful factors
If acoustic noise canceling technique with ambient sound filtering is used, then noise removal capability is improved, but hardware scale and costs increase due to requiring additional microphones
Solution Approach 1:
The patent extracts only the necessary audio signal processing function from the complex acoustic noise canceling system. Instead of using multiple microphones and complex ambient sound processing, it extracts the core filtering function and applies it directly to the input audio signal through software-based frequency band attenuation, achieving noise removal without additional hardware
Solution Approach 2:
The patent replaces the mechanical hardware system (additional microphones and acoustic processing circuitry) with a software-based signal processing system. The filter processing unit implements frequency band attenuation through digital signal processing algorithms, substituting physical hardware with computational methods to achieve the same noise removal effect
2Adaptability or versatility
If manual equalizer adjustment is used, then sound control flexibility is improved, but ease of operation deteriorates due to complexity and lack of pitch sense
Solution Approach 1:
The patent implements self-service by enabling the system to automatically analyze the input audio signal and autonomously determine optimal filter settings. The filter setting unit automatically identifies frequency bands containing noise and applies appropriate attenuation without requiring manual user adjustment, making the system both flexible and easy to operate
Solution Approach 2:
The patent incorporates feedback mechanisms where the system continuously monitors the input audio signal characteristics and automatically adjusts filter settings based on the analyzed signal properties. This closed-loop approach allows the system to adapt to different audio content and noise conditions automatically, providing both flexibility and ease of use
3Manufacturing precision
If filtering is applied to control sound of target audio source, then sound control precision is improved, but delay increases affecting real-time property
Solution Approach 1:
The patent applies partial action by selectively attenuating only the specific frequency bands that contain noise, rather than applying comprehensive heavy filtering across all frequency ranges. This targeted approach maintains sound control precision for the target audio source while minimizing processing complexity and delay
Solution Approach 2:
The patent dynamically changes filtering parameters based on the analyzed input signal characteristics. The filter setting unit adjusts attenuation levels and frequency band selections in real-time according to the detected noise characteristics, optimizing the balance between sound control precision and processing speed for different audio scenarios
Data Source
AI summary
For example, it becomes possible to suppress an increase in hardware scale and costs and to perform signal processing according to an input signal. Provided is an information processing apparatus including a filter processing unit that performs filtering on an input signal, and a filter setting unit that makes a setting of the filtering such that a sound of a target audio source type in the input signal is controlled and the setting is determined using an estimation result obtained from the input signal by an estimation algorithm.