Microphone Array Dereverberation via Adaptive Beamforming
Find Innovative SolutionsGenerate Solutions
Solution Overview
Problem
In vehicle cabins, microphone arrays face challenges in efficiently receiving audio signals while effectively reducing reverberation and noise, particularly from engine sounds, wind, and vibrations, which degrades communication quality.
Innovation Solution
A dereverberation and noise reduction method using a SIMO microphone array that applies beamforming, inverse filtering, blocking matrix processing, and adaptive filtering to isolate and suppress side-lobe portions of audio signals, adjusting parameters based on feedback to enhance output audio quality.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Ease of operation
If the receiving microphone is disposed away from the driver (on the driving wheel or around the windscreen), then the driver can have hands-free operation, but the microphone receives more interfering noises (engine noise, wind noise, vibration noise) and reverberation from the closed vehicle cabin
Solution Approach 1:
The patent divides the audio signal processing into multiple independent processing channels, each handling specific noise types (engine noise, wind noise, vibration noise, reverberation) separately. This segmentation allows targeted processing of different harmful factors while preserving the driver's speech signal.
Solution Approach 2:
The patent introduces reference microphones as intermediary elements that capture only the interfering noises and reverberation without the driver's speech. These reference signals serve as mediators that are processed and subtracted from the main microphone signal to eliminate harmful factors.
2Object-affected harmful factors
If beamforming processing is applied to reduce side-lobe portions, then noise reduction is improved, but the main-lobe portion may be affected and adaptive adjustment is required
Solution Approach 1:
The patent employs adaptive filtering that uses feedback from the processed signal to continuously adjust the beamforming parameters. The system monitors the output and modifies the side-lobe suppression strength to prevent degradation of the main-lobe signal, ensuring optimal performance under varying conditions.
Solution Approach 2:
The beamforming parameters are made dynamic rather than fixed. The system adapts the beamforming weights and suppression levels in real-time based on the acoustic environment and signal characteristics, allowing the main-lobe integrity to be maintained while effectively suppressing side-lobe noise.
3Object-affected harmful factors
If multiple processing steps (beamforming, blocking matrix, adaptive filtering) are applied sequentially, then dereverberation and noise reduction effectiveness is improved, but the processing complexity increases
Solution Approach 1:
The complex processing task is segmented into distinct functional blocks: beamforming processing, blocking matrix processing, and adaptive filtering processing. Each block handles a specific aspect of noise and reverberation reduction, making the overall complex system manageable and allowing independent optimization of each processing stage.
Solution Approach 2:
The patent combines multiple processing techniques (beamforming, blocking matrix, adaptive filtering) into a unified processing framework where the outputs of earlier stages feed into subsequent stages. This merging of multiple methods creates a comprehensive solution that addresses various noise sources and reverberation effects simultaneously.
Data Source
AI summary
A dereverberation and noise reduction method adapted for a microphone array and an apparatus using the same are proposed. The microphone array receives a plurality of audio signals from an audio source. The dereverberation and noise reduction method includes the following steps. The received audio signals are processed by a beamforming processing, and a first audio signal is generated. Besides, the received audio signals are processed by a suppression processing, and a suppression audio vector is generated. Further, suppression audio vector is processed by an adaptive filtering processing, and a second audio signal is generated. In addition, the second audio signal is subtracted from the first audio signal to acquire an audio output signal, where parameters of the adaptive filtering processing are adjusted according to a feedback of the audio output signal.