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US5590241A
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Speech processing system and method for enhancing a speech signal in a noisy environment
Assignee: NXP USA Inc
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Filing: 1993-04-30
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Granted: 1996-12-31
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Expires: 2013-04-30
Abstract
Abstract A speech processing system (30) operates in a noisy environment (20) by performing adaptive prediction between inputs from two sensors positioned to transduce speech from a speaker, such as an accelerometer and a microphone. An adaptive filter (37) such as a finite impulse response (FIR) filter receives a digital accelerometer input signal, adjusts filter coefficients according to an estimation error signal, and provides an enhanced speech signal as an output. The estimation error signal is a difference between a digital microphone input signal and the enhanced speech signal. In one embodiment, the adaptive filter (37) selects a maximum one of a first predicted speech signal based on a relatively-large smoothing parameter and a second predicted speech signal based on a relatively-small smoothing parameter, with which to normalize a predicted signal power. The predicted signal power is then used to adapt the filter coefficients.
Inventors
- 👤 Sangil Park
- 👤 Ed F. Martinez
- 👤 Dae-Hee Youn
Bibliographic Details
Application Number
—
Jurisdiction
US
Data Source
google_patents
Last Synchronized
2026-11-09 14:18
CPC Classifications
G; G10; G10L; G10L15/00; G10L15/20; G10L21/00; G10L21/02; G10L21/0208; G10L21/0216; G10L2021/02161; G10L2021/02165
Claims Analysis
Total Claims: 12
Claim 1
Independent Claim
245 words
A method for enhancing speech signals in a noisy environment, comprising the steps of: inputting a digital speech signal x(k) at an input of a plurality of successive delay elements whose outputs form a like plurality of taps of an adaptive finite impulse response (FIR) filter; inputting said digital speech signal and each of said plurality of taps to corresponding inputs of a plurality of variable multipliers; computing a first signal power estimate y(k) at a sample point k given by the formula y(k)=β 1 y(k-1)+(1-β 1 )x 2 (k); computing a second signal power estimate z(k) at said sample point k given by the formula z(k)=β 2 z(k-1)+(1-β 2 )x 2 (k); choosing a value of β 1 greater than a value of β 2 ; selecting an overall signal power estimate yz(k) at said sample point k as a maximum one of said first signal power estimate y(k) and said second signal power estimate z(k); recursively updating a plurality of FIR filter coefficients corresponding to said plurality of variable multipliers according to a normalized least-mean-squares (NLMS) prediction using said overall signal power estimate yz(k) and an estimation error signal to provide updated values of said plurality of FIR filter coefficients; providing said updated values of said plurality of FIR filter coefficients to coefficient inputs of said plurality of variable multipliers; and summing outputs of said plurality of variable multipliers to provide an enhanced speech signal as an output of said adaptive FIR filter.
Claim 2
Independent Claim
245 words
A method for enhancing speech signals in a noisy environment, comprising the steps of: inputting a digital speech signal x(k) at an input of a plurality of successive delay elements whose outputs form a like plurality of taps of an adaptive finite impulse response (FIR) filter; inputting said digital speech signal and each of said plurality of taps to corresponding inputs of a plurality of variable multipliers; computing a first signal power estimate y(k) at a sample point k given by the formula y(k)=β 1 y(k-1)+(1-β 1 )x 2 (k); computing a second signal power estimate z(k) at said sample point k given by the formula z(k)=β 2 z(k-1)+(1-β 2 )x 2 (k); choosing a value of β 1 greater than a value of β 2 ; selecting an overall signal power estimate yz(k) at said sample point k as a maximum one of said first signal power estimate y(k) and said second signal power estimate z(k); recursively updating a plurality of FIR filter coefficients corresponding to said plurality of variable multipliers according to a normalized least-mean-squares (NLMS) prediction using said overall signal power estimate yz(k) and an estimation error signal to provide updated values of said plurality of FIR filter coefficients; providing said updated values of said plurality of FIR filter coefficients to coefficient inputs of said plurality of variable multipliers; and summing outputs of said plurality of variable multipliers to provide an enhanced speech signal as an output of said adaptive FIR filter.
Claim 3
Dependent Claim
22 words
The method of claim 1 further comprising the step of digitizing an accelerometer input signal to provide said digital speech signal x(k).
Claim 4
Dependent Claim
26 words
The method of claim 2 further comprising the step of subtracting said enhanced speech signal from a second speech signal to provide said estimation error signal.
Claim 5
Dependent Claim
21 words
The method of claim 3 further comprising the step of digitizing a microphone input signal to provide said second speech signal.
Claim 6
Dependent Claim
70 words
The method of claim 1 wherein said step of selecting said overall signal power estimate yz(k) further comprises the step of selecting said overall signal power estimate yz(k) at said sample point k as said maximum one of said first signal power estimate y(k) and said second signal power estimate z(k) when either said first signal power estimate y(k) or said second signal power estimate z(k) exceeds a predetermined threshold.
Claim 7
Dependent Claim
180 words
An adaptive finite impulse response filter, comprising: a signal input for receiving an input signal; a feedback input for receiving an error signal; an output for providing a filtered output signal; a plurality of filter taps coupled in series with said input signal; a plurality of variable multipliers for multiplying said input signal and outputs of said plurality of filter taps by corresponding ones of a plurality of coefficients; a summing device for providing said filtered output signal as a sum of outputs of each of said plurality of variable multipliers; and prediction means for receiving said error signal, and for recursively updating each coefficient according to a maximum one of first and second predicted values of said filtered output signal, in response to said error signal; said first predicted value of said filtered output signal having a first smoothing parameter associated therewith; said second predicted value of said filtered output signal having a second smoothing parameter associated therewith; said first smoothing parameter providing a faster prediction response after a change in said input signal than said second smoothing parameter.
Claim 8
Dependent Claim
180 words
An adaptive finite impulse response filter, comprising: a signal input for receiving an input signal; a feedback input for receiving an error signal; an output for providing a filtered output signal; a plurality of filter taps coupled in series with said input signal; a plurality of variable multipliers for multiplying said input signal and outputs of said plurality of filter taps by corresponding ones of a plurality of coefficients; a summing device for providing said filtered output signal as a sum of outputs of each of said plurality of variable multipliers; and prediction means for receiving said error signal, and for recursively updating each coefficient according to a maximum one of first and second predicted values of said filtered output signal, in response to said error signal; said first predicted value of said filtered output signal having a first smoothing parameter associated therewith; said second predicted value of said filtered output signal having a second smoothing parameter associated therewith; said first smoothing parameter providing a faster prediction response after a change in said input signal than said second smoothing parameter.
Claim 9
Dependent Claim
31 words
The adaptive finite impulse response filter of claim 6 wherein said prediction means provides both said first predicted value and said second predicted value according to a normalized least-mean-squares (NLMS) predictor.
Claim 10
Dependent Claim
195 words
An adaptive finite impulse response filter, comprising: a signal input for receiving an input signal; a feedback input for receiving an error signal; an output for providing a filtered output signal; a plurality of filter taps coupled in series with said input signal; a plurality of variable multipliers for multiplying said input signal and outputs of each of said filter taps by corresponding ones of a plurality of coefficients; a summing device for providing said filtered output signal as a sum of outputs of each of said plurality of multipliers; and a predictor having an input for receiving said error signal, and outputs for providing corresponding ones of said plurality of coefficients; said predictor computing first and second predicted values of said output signal and recursively updating each of said plurality of coefficients as determined by a maximum one of said first and second predicted values of said output signal, and said error signal; said first predicted value characterized as providing a faster prediction response after a change in said input signal than said second predicted value, whereby the adaptive finite impulse response filter is characterized as having a fast attack and a slow release.
Claim 11
Dependent Claim
195 words
An adaptive finite impulse response filter, comprising: a signal input for receiving an input signal; a feedback input for receiving an error signal; an output for providing a filtered output signal; a plurality of filter taps coupled in series with said input signal; a plurality of variable multipliers for multiplying said input signal and outputs of each of said filter taps by corresponding ones of a plurality of coefficients; a summing device for providing said filtered output signal as a sum of outputs of each of said plurality of multipliers; and a predictor having an input for receiving said error signal, and outputs for providing corresponding ones of said plurality of coefficients; said predictor computing first and second predicted values of said output signal and recursively updating each of said plurality of coefficients as determined by a maximum one of said first and second predicted values of said output signal, and said error signal; said first predicted value characterized as providing a faster prediction response after a change in said input signal than said second predicted value, whereby the adaptive finite impulse response filter is characterized as having a fast attack and a slow release.
Claim 12
Dependent Claim
30 words
The adaptive finite impulse response filter of claim 8 wherein said prediction computes both said first predicted value and said second predicted value according to a normalized least-mean-squares (NLMS) predictor.
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USPTO Annuity & Maintenance Windows
| Annuity Stage | Window Opens | Payment Deadline | Surcharge Period Until | Estimated Fee | Status |
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| 3.5 Years (Year 4) | 1999-12-31 | 2000-07-01 | 2000-12-31 | $2000.00 | Window Closed / Lapsed |
| 7.5 Years (Year 8) | 2003-12-31 | 2004-07-01 | 2004-12-31 | $3760.00 | Window Closed / Lapsed |
| 11.5 Years (Year 12) | 2007-12-31 | 2008-07-01 | 2008-12-31 | $7700.00 | Window Closed / Lapsed |