Microphone array signal enhancement
View Patent ↗A system and method facilitating signal enhancement utilizing an adaptive filter is provided. The invention includes an adaptive filter that filters an input based upon a plurality of adaptive coefficients and modifies the adaptive coefficients based on a feedback output. A feedback component provides the feedback output based, at least in part, upon a non-linear function of the acoustic reverberation reduced output. Optionally, the system can further include a linear prediction (LP) analyzer and/or a LP synthesis filter. The system can enhance signal(s), for example, to improve the quality of speech that is acquired by a microphone by reducing reverberation. The system utilizes, at least in part, the principle that certain characteristics of reverberated speech are measurably different from corresponding characteristics of clean speech. The system can employ a filter technology (e.g., reverberation reducing) based on a non-linear function, for example, the kurtosis metric.
1. An audio enhancement system, comprising:
a first adaptive filter that filters an input signal based, at least in part, upon a plurality of adaptive coefficients, the first adaptive filter providing a quality enhanced output;
a second adaptive filter that filters the input signal based, at least in part, upon the plurality of adaptive coefficients, the second adaptive filter modifying at least one of the plurality of adaptive coefficients based, at least in part, upon a feedback output, the second adaptive filter further providing an output; and,
a feedback component that provides the feedback output based, at least in part, upon a non-linear function of the output of the second adaptive filter, the non-linear function being based on maximizing kurtosis of a linear predictive coding (LPC) residual of the quality enhanced output.
2. The audio enhancement system of claim 1 , the second adaptive filter being adapted based on a delayed version of the input signal.
3. The audio enhancement system of claim 1 , the second adaptive filter being adapted based on a modified version of the input signal.
4. The audio enhancement system of claim 1 , the second adaptive filter being adapted based on an equalized version of the input signal.
5. The audio enhancement system of claim 1 , the second adaptive filter running at a different data rate than the first adaptive filter.
6. The audio enhancement system of claim 5 , the second adaptive filter running at a lower data rate than the first adaptive filter.
7. The audio enhancement system of claim 5 , the second adaptive filter running at a higher data rate than the first adaptive filter.
8. The audio enhancement system of claim 1 , the second adaptive filter using data stored in the system.
9. The audio enhancement system of claim 8 , the second adaptive filter using the data stored in the system more than once.
10. The audio enhancement system of claim 8 , the second adaptive filter using the data stored in the system in a different order than the time-of-arrival to a microphone order.
11. The audio enhancement system of claim 1 , the adaptive filter using data modified to edit out non-voiced parts of the input signal.
12. The audio enhancement system of claim 1 , further comprising a plurality of second adaptive filters.
13. The audio enhancement system of claim 12 , the first adaptive filter and the plurality of second adaptive filters employing the same plurality of adaptive coefficients, the plurality of adaptive coefficients being modified, at least in part, based on an output of at least one of the second adaptive filters.
14. The audio enhancement system of claim 13 , adaptation feedback provided by at least one of the second adaptive filters being based on a non-linear function.
15. The audio enhancement system of claim 14 , the non-linear function being based, at least in part, upon selecting a threshold, and providing positive feedback for samples above the threshold and negative feedback for samples below that threshold.