Noise suppression in the frequency domain by adjusting gain according to voicing parameters
View Patent ↗An input signal enters a noise suppression system in a time domain and is converted to a frequency domain. The noise suppression system then estimates a signal to noise ratio of the frequency domain signal. Next, a signal gain is calculated based on the estimated signal to noise ratio and a voicing parameter. The voicing parameter may be determined based on the frequency domain signal or may be determined based on a signal ahead of the frequency domain signal with respect to time. In that event, the voicing parameter is fed back to the noise suppression system, for example, by a speech coder, to calculate the signal gain. After calculating the gain, the noise suppression system modifies the signal using the calculated gain to enhance the signal quality. The modified signal may further be converted from the frequency domain back to the time domain for speech coding.
1. A method of suppressing noise in a signal, said method comprising the steps of:
estimating a signal to noise ratio for said signal;
classifying said signal to a classification;
calculating a gain for said signal using said signal to noise ratio and said classification; and
modifying said signal using said gain;
wherein said calculating step calculates said gain based on γ dh =μ g (σ″ q −σ th )−γ n , wherein μ g is adjusted according to said classification, and wherein γ dh is a gain in a db domain, μ g is a gain slope, σ″ q is a modified signal-to-noise ratio, σ th is a threshold level, and γ n is an overall gain factor.
2. The method of claim 1 further comprising a step of estimating a pitch correlation for said signal, wherein said calculating step further uses said pitch correlation.
3. The method of claim 1 , wherein said signal is one channel of a plurality of channels of a speech signal.
4. The method of claim 2 , wherein μ g is further adjusted according to said pitch correlation.
5. The method of claim 1 , wherein said signal is in a time domain, and said method further comprises a step of converting said signal from said time domain to a frequency time prior to said estimating step.
6. The method of claim 1 , wherein said signal is in a frequency domain, and said method further comprising a step of converting said signal from said frequency domain to a time domain after said modifying step.
7. A method of suppressing noise in a signal having a first signal portion and a second signal portion, wherein said first signal portion is a look-ahead signal of said second signal portion, said method comprising the steps of:
computing a voicing parameter using said first signal portion;
estimating a signal to noise ratio for said second signal portion;
calculating a gain for said second signal portion using said signal to noise ratio and said voicing parameter; and
modifying said signal using said gain;
wherein said calculating step calculates said gain based on γ db =μ g (σ″ q −σ th )+γ n , wherein μ g is adjusted according to said voicing parameter, and wherein γ dh is a gain in a db domain, μ g is a gain slope, σ″ q is a modified signal-to-noise ratio, σ th is a threshold level, and γ n is an overall gain factor.
8. The method of claim 7 , wherein said voicing parameter is computed by a speech coder.
9. The method of claim 7 , wherein said voicing parameter is a speech classification of said first signal portion.
10. The method of claim 7 , wherein said voicing parameter is a pitch correlation of said first signal portion.
11. The method of claim 7 , wherein said signal is in a time domain, and said method further comprises a step of converting said signal from said time domain to a frequency time prior to said estimating step.
12. The method of claim 7 , wherein said signal is in a frequency domain, and said method further comprising a step of converting said signal from said frequency domain to a time domain after said modifying step.
13. A noise suppression system comprising:
a signal to noise ratio estimator;
a signal classifier;
a signal gain calculator; and
a signal modifier;
wherein said estimator estimates a signal to noise ratio of said signal, said signal is given a classification using said signal classifier, said signal gain is calculated based on said signal to noise ratio and said classification using said calculator, and wherein said signal modifier modifies said signal by applying said gain; and
wherein said calculator calculates said gain based on γ db =μ g (σ″ q −σ th )+γ n , wherein μ g is adjusted according to said classification, and wherein γ db is a gain in a db domain, μ g is a gain slope, σ″ q is a modified signal-to-noise ratio, σ th is a threshold level, and γ n is an overall gain factor.
14. The system of claim 13 further comprising a signal pitch estimator for estimating a pitch correlation of said signal for use by said gain calculator.
15. The system of claim 13 further comprising a frequency-to-time converter to convert said signal from a frequency domain to a time domain.
16. A system capable of suppressing noise in a signal having a first signal portion and a second signal portion, wherein said first signal portion is a look-ahead signal of said second signal portion, said system comprising:
a signal processing module for computing a voicing parameter of said first signal portion;
a signal to noise ratio estimator;
a signal gain calculator; and
a signal modifier;
wherein said estimator estimates a signal to noise ratio of said second signal portion, said second signal portion gain is calculated based on said signal to noise ratio and said voicing parameter using said calculator, and wherein said signal modifier modifies said second signal portion by applying said gain; and
wherein said signal gain calculator determines said gain based on γ db =μ g (σ″ q −σ th )+γ n , wherein μ g is adjusted according to said voicing parameter, and wherein μ db is a gain in a db domain, μ g is a gain slope, σ″ q is a modified signal-to-noise ratio, σ th is a threshold level, and γ n is an overall gain factor.
17. The system of claim 16 , wherein said signal processing module is a speech coder.
18. The system of claim 16 , wherein said voicing parameter is a speech classification of said first signal portion.
19. The system of claim 16 , wherein said voicing parameter is a pitch correlation of said first signal portion.
20. The system of claim 16 further comprising a frequency-to-time converter to convert said second signal portion of said signal from a frequency domain to a time domain.