Digital Filter Coefficient Scaling for Transient Minimization
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Solution Overview
Problem
Digital microphones experience audible transients due to gain changes in the signal processing chain, which degrade performance and fail to meet customer requirements for dynamic acoustic overload point (AOP) switching, increased signal-to-noise ratio (SNR), and reduced power consumption.
Innovation Solution
Implementing a filter with delay lines and scaled digital filter coefficients during transient modes to minimize audible transients by temporarily scaling coefficients based on gain changes, using memory components or lookup tables to store and adjust coefficient values.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Adaptability or versatility
If gain change is applied in PGA to achieve higher dynamic range, then dynamic acoustic overload point switching is improved, but audible transients are generated in the digital filter chain
Solution Approach 1:
The patent applies preliminary action by pre-scaling the filter coefficients before the gain change occurs in the PGA. The coefficients are adjusted in advance to compensate for the upcoming gain transition, ensuring that when the gain changes, the filter output remains continuous and transients are avoided. This proactive coefficient adjustment resolves the contradiction by preparing the system ahead of time to handle the gain switching without generating audible artifacts.
Solution Approach 2:
The patent implements parameter changes by dynamically adjusting the filter coefficients based on the gain setting. When the PGA gain changes to achieve different dynamic ranges, the filter coefficients are correspondingly scaled to maintain proper filter behavior. This parameter adaptation allows the system to switch between dynamic acoustic overload points while keeping the filter response continuous, thereby eliminating audible transients.
2Manufacturing precision
If digital compensation is applied to achieve linear signal, then signal linearity is improved, but complex digital processing increases device complexity
Solution Approach 1:
The patent merges the compensation function into the filter coefficient values themselves. Instead of implementing a separate digital compensation block that processes the signal independently, the compensation is embedded directly in the filter coefficients that are applied during normal filtering operations. This integration achieves signal linearity compensation while reducing device complexity by eliminating redundant processing stages.
Solution Approach 2:
The filter performs self-compensation by using its own coefficient scaling mechanism to correct linearity issues. The same filtering operation that processes the audio signal also applies the necessary compensation when coefficients are appropriately scaled. This self-service approach achieves linear signal output without requiring additional dedicated compensation circuitry, thereby reducing overall device complexity.
Data Source
AI summary
A digital filter includes a delay line; coefficients coupled to the delay line; and a summer coupled to at least one of the of coefficients, wherein the coefficients each include a first value during a normal mode of operation, and wherein at least one of the coefficients comprises a second value during a transient mode of operation.


