Adaptive LPC noise reduction system

• US 8,447,044 B2
• Filed: 05/17/2007
• Issued: 05/21/2013
• Est. Priority Date: 05/17/2007
• Status: Active Grant

• Current Assignee: Blackberry Limited
• Original Assignee: QNX Software Systems Limited (Canada) (Blackberry Limited)
• Inventors: Nongpiur, Rajeev, Hetherington, Phillip A.
• Primary Examiner(s): Chin, Vivian
• Assistant Examiner(s): Zhang, Leshui
• Application Number:
  US11/804,577
• Publication Number:
  US 20080285773A1
• Time in Patent Office: 2,196 Days
• Field of Search: 381/94.1, 381/71.1, 381/71.7, 381/71.14, 381/106, 381/317, 381/71.11, 381/71.12, 381/94.7, 381/94.8, 381/302, 381/86, 381/83, 381/92, 381/93, 381/94.2, 381/94.3, 381/97, 381/98, 381/96, 381/99, 381/100, 381/103, 381/121, 381/120, 381/318, 381/320, 367/178, 367/901, 367/124, 367/129, 704/E19.028, 704/207, 704/233, 704/219, 704/262, 700/94, 379/392.01, 379/388.07, 379/406.04, 379/406.05, 379/406.08, 455/501, 455/63.1, 455/67.13, 455/570, 455/114.2, 455/222, 455/278.1, 455/283, 455/296
• US Class Current: 381/71.1
• CPC Class Codes: G10L 21/0208 Noise filtering
• CPC Class Codes: G10L 21/0232 Processing in the frequency domain
• CPC Class Codes: G10L 25/12 the extracted parameters being prediction coefficients

Abstract

A noise suppression system reduces low-frequency noise in a speech signal using linear predictive coefficients in an adaptive filter. A digital filter may update or adapt a limited set of linear predictive coefficients on a sample-by-sample basis. The linear predictive coefficients may be used to provide an error signal based on a difference between the speech signal and a delayed speech signal. The error signal represents an enhanced speech signal having attenuated and normalized low-frequency noise components.

Claims

1. A noise suppression system comprising:

a sampling circuit adapted to sample an audio input signal at a predetermined sampling rate;

a low-pass filter coupled with the sampling circuit and configured to pass low-frequency components of the sampled audio input signal;

a plurality of delay circuits configured to sequentially delay the low-frequency components of the sampled audio input signal to provide sequentially delayed signals;

an adaptive processor configured to process the sequentially delayed signals and update a plurality of linear predictive coefficient (LPC) values on a sample-by-sample basis, based on an error signal, where the error signal is based on a difference between an output of the low-pass filter and an output of the adaptive processor; and

a decision logic device coupled with the adaptive processor and configured to inhibit an update of the LPC values applied to a portion of the sampled audio input signal based on a determination that a wind buffet is not present in the portion of the sampled audio input signal;

a high-pass filter coupled with the sampling circuit and configured to pass high-frequency components of the sampled audio input signal; and

an adder configured to sum the error signal and an output of the high-pass filter to generate an output signal.

2. The system of claim 1, further comprising a conversion circuit configured to convert the output signal to an analog signal as an enhanced output signal having reduced low-frequency components.

3. The system of claim 1, where between 2 and 20 LPC values are updated on a sample-by-sample basis.

4. The system of claim 1, where the error signal represents enhanced sampled audio speech.

5. The system of claim 4, where noise components of the enhanced sampled audio speech are normalized in amplitude, and an average amplitude of the noise components is reduced.

6. The system of claim 1, further comprising a voice activity detector coupled with the decision logic device and configured to detect a presence of a speech signal, where the decision logic device is configured to inhibit updating of the LPC values applied to the speech signal in response to the detected presence of the speech signal.

7. The system of claim 6, where the detection of the speech signal is based on an average energy level of the sampled audio input signal.

8. The system of claim 1, where the low-pass filter passes low-frequency components of the sampled audio input signal to the adaptive processor, and blocks higher-frequency components of the sampled audio input signal.

9. The system of claim 8, where the low-frequency components are flattened in amplitude.

10. The system of claim 1, further comprising a wind buffet detector coupled with the decision logic device and configured to detect whether the wind buffet is present in the portion of the sampled audio input signal, where the decision logic device is configured to inhibit adaptation of the LPC values in response to a determination by the wind buffet detector that the wind buffet is not present.

11. A noise suppression system comprising:

a sampling circuit adapted to sample an input signal at a predetermined sampling rate;

a low-pass filter coupled with the sampling circuit and configured to pass low-frequency components of the sampled input signal;

an adaptive processor coupled with the low-pass filter and configured to update a plurality of linear predictive coefficient (LPC) values on a sample-by-sample basis, based on an error signal;

where the error signal is based on a difference between an output of the low-pass filter and an output of the adaptive processor, and where the LPC values are configured to flatten the error signal across a frequency region of interest to provide the error signal as an enhanced speech signal having reduced low-frequency components;

a wind buffet detector configured to detect whether wind buffets are present in the sampled input signal;

a decision logic device coupled with the wind buffet detector and configured to inhibit adaptation of the LPC values in response to a determination by the wind buffet detector that a wind buffet is not present in the sampled input signal;

a high-pass filter coupled with the sampling circuit and configured to pass high-frequency components of the sampled input signal; and

an adder configured to sum the error signal and an output of the high-pass filter to generate an output signal.

12. The system of claim 11, where the adaptive processor loosely models a human vocal tract.

13. The system of claim 11, where the error signal represents enhanced sampled speech.

14. A method for enhancing a signal provided to a user device, the method comprising:

sampling an audio input signal at a predetermined sample rate;

filtering the sampled audio input signal through a low-pass filter to pass low-frequency components of the sampled audio input signal;

delaying the low-frequency components of the sampled audio input signal by multiple levels of delays to provide sequentially delayed signals;

processing the sequentially delayed signals in an adaptive filter;

adaptively updating linear predictive coefficient (LPC) values on a sample-by-sample basis based on an error signal, where the error signal is based on a difference between an output of the low-pass filter and an output of the adaptive filter;

determining whether a portion of the sampled audio input signal includes wind buffets;

inhibiting an update of the LPC values applied to the portion of the sampled audio input signal in response to a determination that the portion of the sampled audio input signal does not include a wind buffet;

filtering the sampled audio input signal through a high-pass filter to pass high-frequency components of the sampled audio input signal; and

adding the error signal and an output of the high-pass filter to generate an output signal.

15. The method according to claim 14 further comprising converting the output signal to an analog signal and outputting the analog signal as an enhanced signal to the user device.

16. The method according to claim 14, where the adaptive filter loosely models a human vocal tract.

17. The method according to claim 14, where the low-pass filter passes low-frequency components of the sampled audio input signal to the adaptive filter, and blocks higher-frequency components of the sampled audio input signal.

18. The method according to claim 17, where the low-frequency components are flattened in amplitude.

19. A non-transitory computer-readable storage medium having processor executable instructions to provide a noise-reduced signal by performing the acts of:

sampling an audio input signal at a predetermined sample rate;

filtering the sampled audio input signal through a low-pass filter to pass low-frequency components of the sampled audio input signal;

delaying the low-frequency components of the sampled audio input signal by multiple levels of delays to provide sequentially delayed signals;

processing the sequentially delayed signals in an adaptive filter;

adaptively updating linear predictive coefficient (LPC) values on a sample-by-sample basis based on an error signal, where the error signal is based on a difference between an output of the low-pass filter and an output of the adaptive filter;

determining whether a portion of the sampled audio input signal includes wind buffets;

inhibiting an update of the LPC values applied to the portion of the sampled audio input signal in response to a determination that the portion of the sampled audio input signal does not include a wind buffet;

filtering the sampled audio input signal through a high-pass filter to pass high-frequency components of the sampled audio input signal; and

adding the error signal and an output of the high-pass filter to generate an output signal.