Microphone array processing system for noisy multipath environments

• US 20050281415A1
• Filed: 08/05/2005
• Published: 12/22/2005
• Est. Priority Date: 09/01/1999
• Status: Active Grant

• Current Assignee: Northrop Grumman Systems Corporation (Northrop Grumman Corporation)
• Original Assignee: Northrop Grumman Systems Corporation (Northrop Grumman Corporation)
• Inventors: Hsu, Shi-Ping, Lambert, Russell H., Edmonds, Karina L.
• Granted Patent: US 8,000,482 B2
• Time in Patent Office: Days
• US Class Current: 381/92
• CPC Class Codes: H04R 2201/403 Linear arrays of transducers
• CPC Class Codes: H04R 3/005 for combining the signals of two or more microphones specially adapted for hearing aids H04R25/407

Abstract

Apparatus and a corresponding method for processing speech signals in a noisy reverberant environment, such as an automobile. An array of microphones (10) receives speech signals from a relatively fixed source (12) and noise signals from multiple sources (32) reverberated over multiple paths. One of the microphones is designated a reference microphone and the processing system includes adaptive frequency impulse response (FIR) filters (24) enabled by speech detection circuitry (21) and coupled to the other microphones to align their output signals with the reference microphone output signal. The filtered signals are then combined in a summation circuit (18). Signal components derived from the speech signal combine coherently in the summation circuit, while noise signal components combine incoherently, resulting in composite output signal with an improved signal-to-noise ratio. The composite output signal is further processed in a speech conditioning circuit (20) to reduce the effects of reverberation.

Claims

1-10. -10. (canceled)

11. A microphone array processing system for performance enhancement in noisy environments, the system comprising:

a plurality of N microphones positioned to detect speech from a speech source and noise from at least one noise source and to generate corresponding microphone output signals, where N is a positive integer denoting a number of the plurality of microphones, one of the N microphones being designated a reference microphone and the other N−

1 microphones being designated data microphones, the reference microphone and the data microphones receive acoustic signals both from the speech source and from the at least one noise source;

a plurality of adaptive filters, one for each of the data microphones, for aligning each data microphone output signal relative to the reference microphone output signal; and

a signal summation circuit that sums the adaptively filtered microphone output signals with the reference microphone output signal such that signal components resulting from the speech source combine coherently to provide a speech signal having a power gain of approximately N² and such that the signal components resulting from noise combine incoherently to provide a noise signal having power gain of approximately N to produce a corresponding increased signal-to-noise ratio.

12. The system of claim 11, further comprising a plurality of bandpass filters configured to remove a known spectral band containing noise from each of the microphone output signals, the plurality of adaptive filters aligning each of the bandpass filtered output signals from the data microphones relative to the reference microphone output signal.

13. The system of claim 12, wherein the plurality of adaptive filters are updated based on the output signal from the bandpass filter that filters the reference microphone output signal.

14. The system of claim 13, wherein each of the plurality of adaptive filters is configured to update a filter weight value according to a block frequency domain least mean square adaptive update procedure.

15. The system of claim 11, wherein each of the plurality of adaptive filters further comprises a summation circuit that subtracts the output of a respective adaptive filter from the reference microphone output signal to provide a corresponding error signal, each of the plurality of adaptive filters adapting to minimize the corresponding error signal.

16. The system of claim 15, wherein each of the plurality of adaptive filters further comprises a weight vector, representing weighting factors, that is updated based on the corresponding error signal and applied to successive outputs of a tapped delay line of the respective adaptive filter.

17. The system of claim 11, further comprising speech detection circuitry that enables the plurality of adaptive filters in response to detecting speech from the speech source.

18. The system of claim 11, further comprising speech conditioning circuitry that processes the speech output signal to provide a resulting speech signal having an amplitude gain of approximately N.

19. The system of claim 11, wherein each of the plurality of adaptive filters further comprises:

means for filtering data microphone output signals by convolution with a vector of weight values in the frequency domain;

means for comparing the filtered data microphone output signal from one of the data microphones with an output signal from the reference microphone in the frequency domain and deriving therefrom an error signal; and

means for adjusting the weight values convolved with the data microphone output signals in the frequency domain to minimize the error signal.

20. The system of claim 19, wherein each of the adaptive filters further includes Fast Fourier Transform means to transform successive blocks of data microphone output signals to a frequency domain representation to facilitate filtering in the frequency domain.

21. The system of claim 11, wherein each of the plurality of adaptive filters is configured to invert an acoustic path transfer function of the corresponding data microphone and apply a reference acoustic path transfer function of the reference microphone to yield the respective adaptively filtered microphone output signal, such that the signal components resulting from the speech source are added by the signal the summation circuit to produce a speech output signal having a corresponding increased amplitude gain.

22. The system of claim 21, further comprising speech conditioning circuitry that convolves the speech output signal with the reference acoustic path transfer function to provide a resulting speech signal having an amplitude gain of approximately N.

23. A system for improving detection of speech signals, the system comprising:

a plurality of bandpass filters that remove a known spectral band containing noise from a plurality of microphone output signals to provide corresponding bandpass filtered output signals, the plurality of microphone output signals corresponding to acoustic signals both from a speech source and from at least one noise source, one of the plurality of microphone output signals designated a reference microphone signal and the other microphone output signals being data microphone signals;

a plurality of adaptive filters, one for each of the data microphone output signals, that adaptively filter respective bandpass filtered output signals for each of the data microphone output signals and provide adaptively filtered output signals that are aligned relative to the reference microphone signal; and

a signal summation circuit that sums the adaptively filtered output signals such that speech signal contributions from the data microphones are added coherently to provide a speech output signal having an amplitude gain that approximates a number of the signals being summed by the signal summation circuit and such that signal components resulting from noise combine incoherently to provide a noise signal having an amplitude gain of approximately a square root of the number of the signals being summed by the signal summation circuit to produce a corresponding increased signal-to-noise ratio.

24. The system of claim 23, wherein the plurality of adaptive filters are updated based on the bandpass filtered output signal for the reference microphone output signal.

25. The system of claim 24, wherein each of the plurality of adaptive filters further comprises a summation circuit that subtracts the respective adaptively filtered output signal from the bandpass filtered output signal to provide a corresponding error signal, each of the plurality of adaptive filters adapting to minimize the corresponding error signal.

26. The system of claim 25, wherein each of the plurality of adaptive filters further comprises a weight vector, representing weighting factors, that is updated based on the corresponding error signal and applied to successive outputs of a tapped delay line of the respective adaptive filter.

27. The system of claim 23, wherein each of the plurality of adaptive filters is configured to invert an acoustic path transfer function of a corresponding data microphone and apply a reference acoustic path transfer function of the reference microphone to yield each respective adaptively filtered microphone output signal, such that the summation circuit adds the signal components resulting from the speech source to provide the speech output signal having the amplitude gain.

28. The system of claim 27, further comprising speech conditioning circuitry that convolves the speech output signal with the reference acoustic path transfer function to provide a resulting speech output signal having an amplitude gain of approximately N, where N denotes the number of the signals being summed by the signal summation circuit.

29. The system of claim 23, further comprising speech detection circuitry that enables the plurality of adaptive filters in response to detecting speech from the speech source.

30. A method for improving detection of speech signals, the method comprising:

receiving a plurality of microphone output signals from a plurality of microphones positioned to detect speech from a single speech source and noise from at least one noise source, one of the microphone output signals being designated a reference microphone output signal and the others being designated data microphone output signals, wherein plurality of microphone output signals correspond to acoustic signals both from the speech source and from the at least one noise source;

filtering the microphone output signals in a plurality of bandpass filters to eliminate from the microphone output signals a known spectral band containing noise;

adaptively filtering the microphone output signals to align each of the data microphone output signals with the reference microphone output signal; and

combining the adaptively filtered microphone output signals by adding the signal contributions from the speech source coherently to provide a speech amplitude gain that is proportional to a number of signals being added together and by adding the signal components resulting from noise incoherently to provide a noise amplitude gain that is proportional to the square root of the number of signals being added together, whereby a corresponding increased signal-to-noise ratio is produced.

31. The method of claim 30, further comprising conditioning the combined adaptively filtered output signals to reduce reverberation effects in the output signal by modifying the spectrum of the cumulative signal obtained from the signal summation circuit to provide a resulting speech signal having an amplitude gain of approximately N, where denotes the number of signals being added together.