In-Vehicle Speech Processing with Virtual Microphone Directivity
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Solution Overview
Problem
Existing in-vehicle communication systems face challenges in maintaining a high Signal-to-Noise (SN) ratio for speech voice output, leading to reduced audibility and increased reverberation when audio device output mixes with microphone inputs, particularly due to design limitations and increased costs with high-directivity microphones or microphone arrays.
Innovation Solution
An in-vehicle communication support system utilizing a signal processing unit with virtual microphones and adaptive filters to enhance SN ratio by converting microphone inputs into virtual microphone outputs, delaying signals to match phase, and combining them to form a virtual microphone array with higher directivity, thereby improving speech voice output without adding physical microphones.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Measurement precision
If a high-directivity microphone is used to improve SN ratio, then speech voice collection quality improves, but device complexity and design limitation increase
Solution Approach 1:
The patent creates a virtual copy of the first microphone at a different position (virtual microphone) to simulate the acoustic characteristics of a high-directivity microphone without using the actual complex physical structure. This virtual microphone is positioned at the first seat to capture speech with better directionality.
Solution Approach 2:
The patent replaces the mechanical solution of using a physically complex high-directivity microphone with a signal processing approach. By using filters and virtual microphone technology, the system achieves high-directivity effects through digital signal manipulation rather than mechanical microphone design.
2Measurement precision
If a microphone array is used to achieve high directivity, then SN ratio improves, but device complexity and processing load increase
Solution Approach 1:
Instead of adding physical microphones to create an array, the patent creates a virtual copy of the existing first microphone at a different spatial position. This virtual microphone replicates the acoustic field characteristics without requiring additional physical sensors, thereby maintaining system simplicity while achieving array-like directivity.
Solution Approach 2:
The patent extracts and separates the directional information from the multi-microphone array concept and implements it through signal processing on a single physical microphone's output. By taking out the essential function of spatial filtering and implementing it digitally, the system avoids the complexity of multiple physical microphones.
3Adaptability or versatility
If audio device output is mixed with microphone input, then communication support improves, but SN ratio decreases due to reverberation
Solution Approach 1:
The patent extracts and removes the reverberation components from the microphone signal using signal processing techniques. By separating and eliminating the unwanted reverberation and audio device output interference, the system preserves the useful speech signal while removing degrading elements, thereby maintaining both communication support and high SN ratio.
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
The system effectively increases directivity and SN ratio of speech voice output from the speaker while maintaining an efficient configuration and suppressing processing load, enhancing communication clarity without the need for additional microphones.
Implementation Method 1
a filter that converts speech voice of a user in the first seat collected by the second microphone into speech voice of a user in the first seat collected by a virtual microphone positioned closer to the first seat than the second microphone
Implementation Method 2
synthesizing outputs of the plurality of microphones by adjusting delay times so that phases of a target sound match, to improve an SN ratio of the target sound
Data Source
AI summary
A right seat processing unit includes, assuming that a voice of a user in a right front seat is a right front seat voice, a filter HR that converts the right front seat voice being output from a left front seat microphone disposed on a headrest of a left front seat into the right front seat voice collected by a right seat virtual microphone that is a virtual microphone located on a left side of a headrest of the right front seat, a delay unit Z-TR that delays and outputs an output from a right front seat microphone located on a right side of the headrest of the right front seat, a filter WRA that extracts the right front seat voice being output from the filter HR, a filter WRB that extracts the right front seat voice being output from the delay unit Z-TR, and a right adder that adds outputs of the filter WRA and the filter WRB and outputs an added output as a right front seat speech voice signal.