Frequency-Domain Adaptive Noise Control for Vehicles
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Solution Overview
Problem
Existing active noise control systems in vehicles face challenges with high computational effort and time delays, especially in real-time applications, due to the use of adaptive filters in the time or frequency domain, and the inefficiency of single-reference systems in generating coherent noise reduction across multiple locations.
Innovation Solution
A partitioned-block adaptive feedforward filtered-reference least-mean-square (FxLMS) algorithm in the frequency domain is employed with multiple reference sensors and loudspeakers (MIMO/MISO configurations) to generate anti-noise, combined with the Remote Microphone Technique (RMT) for error signal estimation, ensuring real-time capability and reduced computational power.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Reliability
If adaptive filters are used in the time domain for noise reduction, then noise reduction effectiveness is improved, but computational effort increases significantly and real-time processing becomes difficult
Solution Approach 1:
The patent transforms the adaptive filter operation from the time domain to the frequency domain using Fast Fourier Transform (FFT). This parameter change allows the system to maintain noise reduction effectiveness while significantly reducing computational complexity, as frequency-domain convolution becomes simple multiplication of spectral components.
Solution Approach 2:
The patent divides the noise reduction system into multiple independent frequency bands using FFT-based processing. Each frequency component can be processed separately through adaptive filtering, allowing parallel computation and reducing the computational burden on the processor while maintaining overall noise reduction performance.
2Productivity
If block processing is used in the frequency domain for adaptive filtering, then computational effort is reduced, but time delay increases significantly
Solution Approach 1:
The patent implements a fractional delay element that performs preliminary time-shifting of the signal before the adaptive filtering operation. This preliminary action compensates for the inherent time delay in block processing, ensuring that the noise reduction effect is applied at the correct time moment rather than being delayed further.
Solution Approach 2:
The patent uses a flexible block length parameter that can be dynamically adjusted to balance between computational efficiency and time delay. The system adapts the block size based on real-time requirements, allowing optimization of the trade-off between processing efficiency and temporal responsiveness.
3Device complexity
If a single reference signal is used for noise reduction, then system complexity is reduced, but noise reduction effectiveness decreases due to lack of coherent relationship with sound in vehicle interior
Solution Approach 1:
The patent divides the noise measurement task into multiple independent channels by placing separate reference sensors at different locations (e.g., engine block, chassis). Each sensor provides a separate reference signal for its local noise source, allowing the system to handle complex noise environments with multiple independent noise generators without requiring a single overly complex reference system.
Solution Approach 2:
The patent transitions from a single-reference scalar approach to a multi-reference vector approach, adding the dimension of spatial distribution. By measuring noise at multiple locations and using multiple secondary speakers, the system creates a distributed acoustic field that better represents the actual noise environment in the vehicle interior.
4Reliability
If multiple reference sensors and loudspeakers are used for noise reduction, then noise reduction effectiveness is improved, but device complexity increases
Solution Approach 1:
The patent divides the overall noise reduction task into multiple independent MISO (multiple-input single-output) channels, each handling a specific loudspeaker and its corresponding error signal. This segmentation allows the complex MIMO problem to be broken down into simpler, independently processable channels, reducing the computational complexity burden on any single processor.
Solution Approach 2:
The patent creates a modular system where each secondary loudspeaker and its associated error signal form a universal MISO channel that can be processed independently. This multi-functional design allows the same processing architecture to handle multiple noise sources and locations using identical algorithmic blocks, reducing overall system complexity through reuse of proven-effective processing units.
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 achieves effective noise reduction at specific locations within vehicles by improving convergence properties and reducing latency, enabling efficient noise suppression with multiple reference sensors and loudspeakers, while maintaining real-time processing capabilities.
Implementation Method 1
acoustic output means, for example loudspeakers, for the acoustic output of sound signals to suppress the noises
Implementation Method 2
a plurality of sensors for detecting reference signals x[n] correlated with the noises
Implementation Method 3
a plurality of acoustic input means, for example microphones, for detecting acoustic error signals m[n], wherein the acoustic input means are arranged in the vicinity of the defined location
Data Source
Figure 1~2
Figure 3~5(b)
Figure 4
AI summary
The present disclosure relates to a device for suppressing noise at a defined location, comprising: a plurality of sensors 6 for detecting reference signals correlated with the noise; a plurality of acoustic output means 7 for the acoustic output of sound signals for suppressing the noise, wherein the acoustic output means are arranged in the vicinity of the defined location; a plurality of acoustic input means 8 for detecting error signals, wherein the acoustic input means are arranged in the vicinity of the defined location; a processing unit 2 which receives and processes the reference signals and the error signals and, based on these, generates control signals for the acoustic output means and outputs them to them;wherein the processing unit generates the control signals by means of adaptive control signal filters 10, and the adaptation of the control signal filters 10 is carried out by means of a method for reducing an error function based on the error signals; wherein at least one control signal is generated based on a plurality of reference signals; wherein at least one control signal filter 10 generates a control signal based on processing in the frequency domain, and the at least one control signal filter 10 is implemented by a plurality of parallel-arranged control signal sub-filters 10', each of which is adapted separately from one another in the frequency domain.