Audio Enhancement via Adaptive Blocking Matrix
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Solution Overview
Problem
The existing beamforming algorithms over-suppress sounds from non-target directions, which is not suitable for applications like hearing aids that aim to simulate the sound collection effect of the human ear.
Innovation Solution
An audio enhancement method that uses a microphone array to generate audio collection signals, performs delay-and-sum and blocking-matrix processing, and applies adaptive filtering with a time-varying attenuation function to control sound attenuation in different directions and frequencies.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Measurement precision
If a typical beamforming algorithm is used to retain sound from a certain direction, then the sound quality in the target direction is improved, but all sounds from other directions are excessively reduced
Solution Approach 1:
The patent applies dynamics by making the attenuation function time-varying rather than static. The attenuation function is updated at predetermined intervals based on changing audio environments, allowing the system to adaptively adjust the degree of suppression for non-target directions. This dynamic adjustment prevents excessive and fixed suppression, enabling the system to maintain target direction sound quality while reducing harmful over-suppression of other sounds.
Solution Approach 2:
The patent changes parameters by introducing a time-varying attenuation function with adjustable update intervals and forgetting factors. These parameter changes allow flexible control over the suppression characteristics - the system can adjust how quickly the attenuation function adapts to environmental changes and how much weight is given to historical data versus new information, thereby balancing target direction enhancement with preservation of non-target sounds.
2Device complexity
If a fixed beamforming algorithm is applied, then the algorithm simplicity is maintained, but the adaptability to different audio environments is reduced
Solution Approach 1:
The patent applies periodic action by updating the attenuation function at predetermined intervals rather than continuously or statically. This periodic update mechanism provides a balance between computational simplicity and environmental adaptability - the system periodically reconfigures itself based on current audio conditions while maintaining relative simplicity between updates. The periodic nature allows the algorithm to adapt to changing environments without requiring constant complex recalculations.
Solution Approach 2:
The patent implements feedback through the adaptive update mechanism where the attenuation function is continuously refined based on incoming audio signals. The system uses feedback from the audio environment to adjust the attenuation characteristics, incorporating a forgetting factor that balances historical performance with new information. This feedback loop enables the algorithm to adapt to different audio environments while maintaining a relatively simple structure through iterative refinement rather than complex redesign.
Data Source
AI summary
Disclosed in the present application are an audio enhancement method and apparatus, and a computer storage medium. The method comprises: generating a group of audio collection signals by means of a microphone array; performing delay-and-sum processing on the group of audio collection signals, so as to generate a delay-and-sum signal; performing blocking matrix processing on the group of audio collection signals, so as to generate a blocking matrix signal; using an adaptive filtering matrix to filter the blocking matrix signal, and removing the filtered blocking matrix signal from the delay-and-sum signal, so as to obtain an enhanced audio output signal. The adaptive filtering matrix is based on at least one attenuation function, and each of the at least one attenuation function is updated at an interval of a corresponding predetermined update interval T.