Method and system for adaptive speech enhancement using frequency specific signal-to-noise ratio estimates

• US 6,098,038 A
• Filed: 09/27/1996
• Issued: 08/01/2000
• Est. Priority Date: 09/27/1996
• Status: Expired due to Fees

• Current Assignee: Oregon Health & Science University
• Original Assignee: Oregon Graduate Institute Of Science & Technology
• Inventors: Avendano, Carlos M., Hermansky, Hynek
• Primary Examiner(s): Hudspeth, David R.
• Assistant Examiner(s): Opsasnick, Michael N.
• Application Number:
  US08/722,547
• Time in Patent Office: 1,404 Days
• Field of Search: 704/201, 704/224, 704/225, 704/226, 704/227, 704/228
• US Class Current: 704/226
• CPC Class Codes: G10L 21/0208 Noise filtering

Abstract

A method and system for adaptively filtering a speech signal in order to suppress noise in the signal. The method includes decomposing the signal into multiple frequency subbands, each having a center frequency, estimating a signal-to-noise ratio for each subband, and providing multiple filters, each filter designed for one of a number of selected signal-to-noise ratio independent of the center frequencies of the subbands. The method also includes selecting a filter for filtering each subband, where the filter selected depends on the signal-to-noise ratio estimated for the subband, filtering each subband according to the filter selected, and combining the filtered subbands to provide an estimated filtered speech signal. The system includes appropriate hardware and software for performing the method.

Claims

1. A method for adaptively filtering a speech signal to suppress noise therein, the method comprising:

decomposing the speech signal into a plurality of frequency subbands, each subband having a center frequency;

estimating a signal-to-noise ratio for each subband;

providing a plurality of filters, each filter designed for one of a plurality of selected signal-to-noise ratios independent of the center frequencies of the plurality of subbands;

selecting one of the plurality of filters for each subband, wherein the filter selected depends on the signal-to-noise ratio estimated for the subband;

filtering each subband according to the filter selected; and

combining the filtered subbands to provide an estimated filtered speech signal.

2. The method of claim 1 wherein decomposing the signal into a plurality of frequency subbands comprises performing a short-time Fourier transform on the signal.

3. The method of claim 2 wherein decomposing the signal into a plurality of frequency subbands further comprises computing a magnitude of each subband and a signal phase.

4. The method of claim 3 wherein estimating a signal-to-noise ratio for each subband comprises computing a histogram of the subband magnitudes.

5. The method of claim 1 wherein providing a plurality of filters comprises computing each filter based on parallel recordings of a clean speech signal and a noisy speech signal.

6. The method of claim 5 wherein providing a plurality of filters comprises:

decomposing the noisy speech signal into a plurality of frequency subbands;

determining a magnitude response at every subband for the plurality of selected signal-to-noise ratios; and

averaging the magnitude responses determined for each one of the plurality of selected signal-to-noise ratios.

7. The method of claim 6 wherein each of the plurality of filters comprises a finite impulse response filter.

8. The method of claim 7 wherein the plurality of filters comprises a filter bank.

9. The method of claim 3 further comprising:

compressing the magnitude of each subband prior to filtering; and

de-compressing the magnitude of each subband after filtering.

10. A system for adaptively filtering a speech signal to suppress noise therein, the system comprising:

means for decomposing the speech signal into a plurality of frequency subbands, each subband having a center frequency;

means for estimating a signal-to-noise ratio for each subband;

a plurality of filters for filtering the subbands, each filter designed for one of a plurality of selected signal-to-noise ratios independent of the center frequencies of the plurality of subband;

means for selecting one of the plurality of filters for each subband, wherein the filter selected depends on the signal-to-noise ratio estimated for the subband; and

means for combining the filtered subbands to provide an estimated filtered speech signal.

11. The system of claim 10 wherein the means for decomposing the signal into a plurality of frequency subbands comprises means for performing a short-time Fourier transform on the signal.

12. The system of claim 11 wherein the means for decomposing the signal into a plurality of frequency subbands further comprises means for computing a magnitude of each subband and a signal phase.

13. The system of claim 12 wherein the means for estimating a signal-to-noise ratio for each subband comprises means for computing a histogram of the subband magnitudes.

14. The system of claim 10 further comprising means for computing the plurality of filters based on parallel recordings of a clean speech signal and a noisy speech signal.

15. The system of claim 14 wherein the means for computing the plurality of filters comprises:

means for decomposing the noisy speech signal into a plurality of frequency subbands;

means for determining a magnitude response at every subband for the plurality of selected signal-to-noise ratios; and

means for averaging the magnitude responses determined for each one of the plurality of selected signal-to-noise ratios.

16. The system of claim 15 wherein each of the plurality of filters comprises a finite impulse response filter.

17. The system of claim 16 wherein the plurality of filters comprises a filter bank.

18. The system of claim 12 further comprising:

means for compressing the magnitude of each subband prior to filtering; and

means for de-compressing the magnitude of each subband after filtering.