Adaptive IIR Filter Coefficient Library for Low-Power ANC
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Solution Overview
Problem
Existing active noise cancellation (ANC) systems in personal audio devices face challenges with high power consumption due to the need for high sampling rates and large filter coefficients in infinite impulse response (IIR) filters, which can lead to instability and require careful design to match magnitude and phase responses within specific frequency ranges.
Innovation Solution
An integrated circuit with an adaptive IIR filter and a coefficient control block that selects coefficients from a library to minimize ambient noise, using a two-microphone system to measure ambient and output signals, and an offline method to create a library of filter coefficients that optimize the filter response for effective noise cancellation.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Reliability
If adaptive filters use high sampling rates to minimize latency, then noise cancellation performance is improved, but power consumption increases
Solution Approach 1:
The patent changes the sampling rate parameter to 1.536 MHz (higher than conventional rates) while using IIR filters with carefully designed coefficients to achieve the desired frequency response. This allows the system to maintain high noise cancellation performance while managing power consumption through efficient filter design rather than simply increasing sampling rate
Solution Approach 2:
The patent replaces conventional FIR (Finite Impulse Response) filter mechanisms with IIR (Infinite Impulse Response) filter mechanisms. The IIR filter structure achieves the same noise cancellation function with fewer coefficients and lower computational complexity, thereby reducing power consumption while maintaining performance
2Reliability
If IIR filter order is increased to improve noise cancellation, then filtering effectiveness is improved, but stability decreases
Solution Approach 1:
The patent carefully adjusts the IIR filter coefficients and filter order parameters to achieve the optimal balance between noise cancellation effectiveness and stability. The coefficients are designed to provide the desired frequency response in the human-perceptible audio bandwidth (100 Hz to 3 KHz) while maintaining system stability through proper pole-zero placement
Solution Approach 2:
The patent implements an adaptive feedback mechanism where the filter coefficients are continuously adjusted based on the error signal from the error microphone. This adaptive feedback allows the filter to optimize its performance while maintaining stability through controlled adaptation of the coefficients
3Loss of time
If filter coefficients are increased to reduce latency, then noise cancellation response is improved, but power consumption increases
Solution Approach 1:
The patent substitutes FIR filter structure with IIR filter structure, which achieves the same latency reduction goal with fewer coefficients. The IIR filter's recursive structure provides longer impulse response with fewer taps, thereby reducing the computational load and power consumption while maintaining low latency
4Adaptability or versatility
If adaptive filtering is used to adapt to environmental changes, then noise cancellation adaptability is improved, but device complexity increases
Solution Approach 1:
The patent uses adaptive algorithms that adjust filter coefficients by changing parameters based on environmental conditions. The coefficient control block modifies the IIR filter coefficients in real-time based on feedback from the error microphone, enabling the system to adapt to different acoustic environments without requiring complex redesign
Solution Approach 2:
The patent implements a dynamic filter system where the coefficients are not fixed but change adaptively in response to environmental conditions. The coefficient control block continuously updates the filter parameters to match the current acoustic environment, providing dynamic adaptation while maintaining manageable complexity through the use of established adaptive filtering techniques
Data Source
AI summary
An integrated circuit for implementing at least a portion of a personal audio device may include an output for providing a signal to a transducer including both a source audio signal for playback to a listener and an anti-noise signal for countering the effects of ambient audio sounds in an acoustic output of the transducer, a reference microphone input for receiving a reference microphone signal indicative of the ambient audio sounds, an error microphone input for receiving an error microphone signal indicative of the output of the transducer and the ambient audio sounds at the transducer, and a processing circuit configured to implement an adaptive infinite impulse response filter having a response that generates the anti-noise signal to reduce the presence of the ambient audio sounds at the error microphone and implement a coefficient control block that shapes the response of the adaptive infinite impulse response filter in conformity with the error microphone signal by generating coefficients that determine the response of the adaptive infinite impulse response filter in order to minimize the ambient audio sounds at the error microphone, wherein the coefficient control block selects the coefficients from a library of filter entries, each filter entry of the library of filter entries defining a respective response for the adaptive infinite impulse response filter.