SYSTEM AND METHOD FOR SPATIAL NOISE SUPPRESSION BASED ON PHASE INFORMATION

• US 20120093338A1
• Filed: 08/08/2011
• Published: 04/19/2012
• Est. Priority Date: 10/18/2010
• Status: Active Grant

• Current Assignee: Arlington Technologies, LLC (Dominion Harbor Enterprises, LLC)
• Original Assignee: Avaya Incorporated
• Inventors: Teutsch, Heinz, LEVI, Avram
• Granted Patent: US 8,913,758 B2
• Time in Patent Office: Days
• US Class Current: 381/92
• CPC Class Codes: H04R 2201/401 2D or 3D arrays of transducers
• CPC Class Codes: H04R 2201/403 Linear arrays of transducers
• CPC Class Codes: H04R 2201/405 Non-uniform arrays of transducers or a plurality of uniform arrays with different transducer spacing
• CPC Class Codes: H04R 2430/23 Direction finding using a sum-delay beam-former
• CPC Class Codes: H04R 3/005 for combining the signals of two or more microphones specially adapted for hearing aids H04R25/407

Abstract

Disclosed herein are systems, methods, and non-transitory computer-readable storage media for suppressing spatial noise based on phase information. The method transforms audio signals to frequency-domain data and identifies time-frequency points that have a parameter (e.g., signal-to-noise ratio) above a threshold. Based on these points, unwanted signals can be attenuated the desired audio source can be isolated. The method can work on a microphone array that includes two microphones or more.

Claims

1. A method comprising:

receiving a first audio signal via a first microphone, and a second audio signal via a second microphone, wherein the first audio signal and the second audio signal are from an audio space including at least one audio source;

performing a short-time Fourier transform of the first audio signal and the second audio signal to yield frequency-domain data;

identifying, in the frequency-domain data, time-frequency points having a parameter above a threshold; and

generating an audio signal based at least in part on the time-frequency points.

2. The method of claim 1, wherein generating the audio signal further comprises applying an inverse short-time Fourier transform.

3. The method of claim 1, wherein generating the audio signal further comprises attenuating time-frequency points having a parameter below the threshold.

4. The method of claim 1, wherein the first audio signal and the second audio signal each represent part of the audio space at a same time.

5. The method of claim 4, wherein the audio space is one of a two-dimensional audio space and a three-dimensional audio space.

6. The method of claim 1, further comprising receiving at least one additional audio signal, and wherein the short-time Fourier transform incorporates the at least one additional audio signal.

7. The method of claim 1, further comprising:

forming a delay-and-sum beamformer with the first microphone and the second microphone; and

aiming the delay-and-sum beamformer at a desired audio source.

8. The method of claim 7, wherein forming the delay-and-sum beamformer further comprises time-aligning the first microphone and the second microphone such that, when the first audio signal and the second audio signal are added, the desired audio source is added coherently and other audio sources are added incoherently.

9. The method of claim 1, wherein identifying the time-frequency points is based on a first phase of the first audio signal and a second phase of the second audio signal.

10. The method of claim 1, wherein identifying the time-frequency points further comprises smoothing the frequency-domain data.

11. The method of claim 10, wherein smoothing the frequency-domain data further comprises applying a time-frequency averaging filter.

12. The method of claim 10, wherein smoothing the frequency-domain data further comprises applying a sliding frequency window and identifying a minimum value in the sliding frequency window.

13. The method of claim 1, wherein the at least one audio source comprises a plurality of separate audio sources, and wherein performing the short-time Fourier transform occurs in parallel for each of the plurality of separate audio sources.

14. The method of claim 1, wherein the parameter is a signal-to-noise ratio.

15. A system comprising:

a processor;

a first microphone;

a second microphone;

a first module configured to control the processor to receive a first audio signal via the first microphone, and a second audio signal via the second microphone, wherein the first audio signal and the second audio signal originate from an audio space including at least one audio source;

a second module configured to control the processor to establish a search pattern of regions in the audio space;

a third module configured to control the processor to perform a short-time Fourier transform of the first audio signal and the second audio signal for each of the regions in the audio space to yield scanned frequency-domain data;

a fourth module configured to control the processor to identify, in the scanned frequency-domain data, a time-frequency point having a highest signal-to-noise ratio; and

a fifth module configured to control the processor to mark as a desired audio source a region in the audio space corresponding to the time-frequency point having the highest signal-to-noise ratio.

16. The system of claim 15, further comprising:

a sixth module configured to control the processor to generate a reconstructed audio signal of the desired audio source from the first audio signal and the second audio signal based on the time-frequency point having the highest signal-to-noise ratio.

17. The system of claim 15, wherein the fourth module is further configured to control the processor to identify the time-frequency point having the highest signal-to-noise ratio for a desired audio signal type.

18. A non-transitory computer-readable storage medium storing instructions which, when executed by a computing device, cause the computing device to isolate an audio source from a particular direction, the instructions comprising:

forming a delay-and-sum beamformer using a first microphone and a second microphone;

aiming the delay-and-sum beamformer at the audio source to receive a first audio signal via the first microphone, and a second audio signal via the second microphone, wherein the first audio signal and the second audio signal include the audio source, to yield a short-time Fourier transform of the first audio signal and the second audio signal;

generating frequency-domain data based on the short-time Fourier transform;

identifying, in the frequency-domain data, time-frequency points having a signal-to-noise ratio above a threshold for the audio source; and

isolating a desired audio signal of the audio source by retaining the time-frequency points having the signal-to-noise ratio above the threshold and attenuating all other time-frequency points.

19. The non-transitory computer-readable storage medium of claim 18, wherein aiming the delay-and-sum beamformer further comprises steering the delay-and-sum beamformer to locations adjacent to the audio source, whereby a wider-range spatial suppression is achieved.

20. The non-transitory computer-readable storage medium of claim 18, wherein forming the delay-and-sum beamformer further comprises time-aligning the first microphone and the second microphone such that, when the first audio signal and the second audio signal are added, the audio source is added coherently and other audio sources are added incoherently.