Table Microphone Array with Non-Linear Beamforming for Hearing Assistance
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Solution Overview
Problem
Current hearing assistance systems face challenges in noisy environments with multiple talkers, as omnidirectional microphones capture unwanted speech and reverberation, leading to reduced speech understanding and the need for manual adjustment of directional microphones.
Innovation Solution
A system with a table microphone unit featuring a non-linear arrangement of microphones generating multiple fixed acoustic beams, where one beam is selected based on signal-to-noise ratio (SNR) and voice activity, and a smooth transition is implemented between beams to avoid noise cancellation drawbacks and manual adjustments.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Adaptability or versatility
If omnidirectional microphones are used to capture speech in noisy environments, then all talkers can be captured, but unwanted speech and reverberation are also captured reducing speech understanding
Solution Approach 1:
The patent divides the omnidirectional sound capture into multiple directional acoustic beams, each focused on a specific talker. The beamformer unit creates separate beams that can be independently selected and combined, allowing the system to capture multiple talkers while excluding unwanted speech and reverberation from other directions.
Solution Approach 2:
The patent applies different acoustic properties to different spatial regions by creating directional beams with specific pickup patterns. Each beam has optimized sensitivity in its target direction while rejecting sound from other directions, providing local quality enhancement for speech understanding without compromising the ability to capture multiple talkers.
2Reliability
If directional microphones are used to focus on a specific talker, then speech understanding is improved, but manual adjustment is required when the preferred talker changes
Solution Approach 1:
The patent implements automatic beam selection and switching based on voice activity detection and signal-to-noise ratio analysis. The system autonomously determines which beam to activate based on current acoustic conditions, eliminating the need for manual adjustment while maintaining optimal speech understanding quality as talkers change.
Solution Approach 2:
The patent creates a dynamic beam switching system that adapts to changing acoustic environments in real-time. Multiple fixed beams are prepared in advance, and the system dynamically selects and transitions between them based on detected voice activity and SNR metrics, providing automatic adaptation without manual intervention.
3Speed
If abrupt beam switching is used to change the active acoustic beam, then response time is fast, but loss of speech portions and unpleasant transitions occur
Solution Approach 1:
The patent implements periodic cross-fading between beams with a controlled transition duration. Instead of instantaneous switching, the system gradually transitions from one beam to another over a defined period, maintaining audio quality continuity while still providing relatively fast response to changing speech conditions.
Solution Approach 2:
The patent prepares multiple beams in advance and maintains them ready for selection. During transitions, the system cushiones the switching process by gradually blending beams, preventing abrupt changes that would cause speech loss or unpleasant artifacts while maintaining readiness to respond to new speech conditions.
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 enhances speech understanding by minimizing unwanted speech and reverberation, maintaining clear audio quality without the need for manual beam direction adjustments, and prevents loss of speech portions or unpleasant transitions.
Implementation Method 1
a beamformer unit for generating a plurality of acoustic beams having a fixed direction, wherein one of the acoustic beams is selected as a presently active beam
Data Source
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AI summary
There is provided a system for providing hearing assistance to a user, comprising: a table microphone unit (10) for capturing audio signals from a speaker's voice, comprising a microphone arrangement (16) comprising at least three microphones (M1, M2, M3) arranged in a non-linear manner, a beamformer unit (48) comprising a plurality of beamformers (BF1, BF2,... ), wherein each beamformer is configured to generate an acoustic beam (B1, B2,...) by beamforming processing of audio signals captured by a subset of the microphones in such a manner that the acoustic beam has a fixed direction, an audio signal analyzer unit (52) for analyzing the beams in order to determine at least one acoustic parameter for each acoustic beam, a beam selection unit (54) for selecting one of the acoustic beams as the presently active beam based on the values of the at least one acoustic parameter, an output unit (60) for providing an acoustic output stream (26), wherein the output unit is configured to provide, during stationary phases of the beam selection, the presently active beam as the output stream, and to provide, during a transition period starting upon switching of the beam selection from a first beam to a second beam, a mixture of the first and second beam with a time-variable weighting of the first and second beam as the output stream so as to enable a smooth transition from the first beam to the second beam during the transition period, a transmission unit (20) for transmitting an audio signal corresponding to the output stream via a wireless link (14); and a hearing assistance device (12) to be worn by the user, comprising a receiver unit (30) for receiving audio signals transmitted from the transmitter of the table microphone unit and an output transducer (40) for stimulation of the user's hearing according to the received audio signals.