Audio Processing for Same-Voice Interference Reduction
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Solution Overview
Problem
In multipoint communication systems, particularly in online conferences, users at the same point experience discomfort due to direct and delayed voice inputs from multiple participants, impairing conversation comfort.
Innovation Solution
An audio processing system with a first and second input interface and a signal processing circuit that reduces the component of a voice signal when both signals are from the same person and clear, adjusting volume and noise cancellation based on clarity and identity determinations.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Adaptability or versatility
If voice signals from multiple participants are output to all users in a multipoint communication system, then communication coverage is improved, but conversation comfort deteriorates due to direct and delayed voice interference
Solution Approach 1:
The system applies different processing to voice signals based on the spatial relationship between users. When a user is physically close to another user, the system prioritizes direct voice transmission to that specific user, while suppressing broadcast to other remote users. This localized quality adjustment resolves the contradiction by providing targeted communication coverage rather than uniform broadcast.
Solution Approach 2:
The system dynamically adjusts voice signal processing based on real-time detection of user proximity and voice clarity. The control unit continuously monitors whether users are close to each other and adjusts the output strategy accordingly - switching between direct transmission mode (when users are close) and broadcast mode (when users are distant). This dynamic adaptation resolves the contradiction between coverage and interference.
2Object-affected harmful factors
If direct voice transmission is prioritized for close users, then conversation comfort is improved, but communication coverage deteriorates for distant users
Solution Approach 1:
The system dynamically switches between two communication modes based on detected user proximity. When users are detected to be close, the system uses direct transmission mode for optimal conversation comfort. When users are distant, it switches to broadcast mode to ensure coverage. This dynamic mode switching resolves the contradiction by adapting the communication strategy to the specific spatial context.
Solution Approach 2:
The system uses feedback from voice clarity detection and proximity sensing to adjust its transmission strategy. The control unit receives information about user positions and voice signal quality, then uses this feedback to determine whether to prioritize direct transmission or broadcast coverage. This feedback loop enables the system to resolve the contradiction adaptively based on actual communication conditions.
3Object-affected harmful factors
If voice signals are processed to reduce components from the same person, then conversation comfort is improved, but signal processing complexity increases
Solution Approach 1:
The system uses each user's own microphone to capture their direct voice, which then serves as a reference signal for suppressing the same user's voice in broadcast transmissions. The user's own voice recording becomes the tool for eliminating their voice from the output, creating a self-service mechanism that reduces self-voice interference without requiring complex external processing.
Solution Approach 2:
The system extracts the direct voice component from the mixture of voice signals using simple comparison logic. By taking out the directly captured voice signal and subtracting it from the processed output, the system removes self-voice interference with minimal processing complexity, resolving the contradiction between interference reduction and processing complexity.
Data Source
AI summary
An audio processing system includes a first input I/F; a second input I/F; and a processor. The first input I/F obtains a first voice signal via a communication line. The second input I/F obtains a second voice signal based on a voice collected by a microphone. The processor outputs an output voice signal based on the first voice signal and the second voice signal to a loudspeaker. The processor causes the output voice signal to include a signal obtained by reducing a component corresponding to the first voice signal, when a first condition and a second condition are met. The first condition is that both the first voice signal and the second voice signal include a voice signal based on a voice uttered by the same person. The second condition is that the second voice signal is clear.