Dynamic noise reduction using linear model fitting

1. A system that improves speech quality comprising:

a spectral converter that is configured to digitize and convert a time varying signal into the frequency domain;

a background noise estimator configured to measure a background noise that is present in the time varying signal detected by a receiver;

a spectral separator in communication with the spectral converter and the background noise estimator that is configured to divide a power spectrum of a speech segment;

a modeler in communication with the spectral separator that fits a plurality of substantially linear functions to differing portions of the speech segment, where the modeler is configured to fit a first line of the plurality of substantially linear functions to a first frequency portion of the speech spectrum and a second line of the plurality of substantially linear functions to a second frequency portion of the speech spectrum;

a dynamic noise adjuster programmed to designate the spectral magnitude of a noisy portion of the speech segment by designating a dynamic adjustment factor that corresponds to the noisy portion of the speech segment, where the dynamic noise adjuster is programmed to calculate a difference in slope between the first line and the second line, and calculate the dynamic adjustment factor based on the difference in slope; and

a dynamic noise processor programmed to attenuate a portion of the noise detected in one or more portions of the speech segment based on the dynamic adjustment factor.

2. The system that improves speech quality of claim 1 where the modeler is configured to approximate a plurality of linear relationships.

3. The system that improves speech quality of claim 2 where the first frequency portion comprises a medium to low frequency portion of an aural spectrum below a predetermined frequency and the second frequency portion comprises a high frequency portion of the aural spectrum above the predetermined frequency.

4. A speech enhancement system that adapts to changing noise conditions heard in a vehicle, comprising:

a time-to-frequency converter that converts portions of a first speech segment into frequency bands;

a signal detector configured to measure a signal power of the frequency bands of the first speech segment;

a background noise estimator configured to measure an aural background noise detected within a vehicle; and

a dynamic noise reduction controller configured to dynamically model the aural background noise in the vehicle to render an attenuated speech segment through a dynamic attenuation of a portion of the noise that occurs in a low frequency portion of the spectrum of the first speech segment below a predetermined frequency;

where the dynamic noise reduction controller is configured to fit a first line to the low frequency portion of the spectrum of the first speech segment, fit a second line to a high frequency portion of the spectrum of the first speech segment above the predetermined frequency, calculate a difference in slope between the first line and the second line, and calculate an attenuation amount for the dynamic attenuation based on the difference in slope.

5. The speech enhancement system of claim 4 further comprising an analog-to-digital converter configured to convert the analog speech segment into a digital signal.

6. The speech enhancement system of claim 5 where the time-to-frequency converter comprises a Short-Time-Fourier-Transform controller.

7. The speech enhancement system of claim 6 where the background noise estimator comprises a power detector configured to average acoustic power in each of the frequency bands.

8. The speech enhancement system of claim 7 further comprising a transient detector configured to disable the background noise estimator when the measured background noise exceeds a predetermined threshold.

9. The speech enhancement system of claim 8 where the dynamic noise reduction controller is configured to discriminate between two or more intervals of a frequency spectrum.

10. The speech enhancement system of claim 8 where the dynamic noise reduction controller is programmed to attenuate a portion of the noise that occurs in a portion of the spectrum of a speech segment.

11. The speech enhancement system of claim 8 where the dynamic noise reduction controller is configured to apply a substantially uniform suppression when a frequency of the speech segment is substantially equal to or greater than the predetermined frequency.

12. The speech enhancement system of claim 11 where the dynamic noise reduction controller is configured to apply a variable suppression to a frequency bin of the speech segment when the frequency bin is less than the predetermined frequency.

13. The speech enhancement system of claim 8 further comprising a wind suppression system in communication with the dynamic noise reduction controller that suppresses the noise generated by moving air.

14. A system that dynamically controls the attenuation gain applied to a signal recorded in a vehicle, comprising:

a power processor configured to measure the signal power in a sound segment in real-time;

a background noise processor configured to measure the background noise detected in the sound segment in real-time;

a dynamic noise reduction processor configured to model the measured background noise by processing multiple linear relationships; and

a dynamic noise suppression filter having a noise suppression gain adjusted in response to the model of the measured background noise;

where the dynamic noise reduction processor is configured to fit a first line to a first frequency portion of the sound segment, fit a second line to a second frequency portion of the sound segment, calculate a difference in slope between the first line and the second line, and calculate the noise suppression gain based on the difference in slope.

15. A system that dynamically controls the attenuation gain applied to a signal of claim 14 , where the first frequency portion comprises a low frequency portion of the sound segment below a predetermined frequency.

16. A system that dynamically controls the attenuation gain applied to a signal of claim 15 , where the second frequency portion comprises a high frequency portion of the sound segment above the predetermined frequency.

17. A method that improves speech quality and intelligibility of a speech segment, comprising:

converting a sound segment into separate frequency bands where each band identifies an amplitude and a phase;

estimating the background noise of a signal by averaging the acoustic power measured in each frequency band;

discriminating between a high portion of the frequency spectrum above a predetermined frequency and a low portion of the frequency spectrum below the predetermined frequency;

fitting a first line to the low portion of the frequency spectrum;

fitting a second line to the high portion of the frequency spectrum;

calculating a difference in slope between the first line and the second line;

modeling a background noise spectrum by determining a substantially constant attenuation to be applied to the high frequency portion of the spectrum and a variable attenuation to be applied to the low portion of the frequency spectrum;

calculating a level for the variable attenuation based on the difference in slope; and

attenuating portions of the background noise from the sound segment by applying the constant attenuation and the variable attenuation.

18. The method that improves speech quality and intelligibility of a speech segment of claim 17 further comprising designating a predetermined frequency band that designates the separation between the high portion of the frequency spectrum and the low portion of the frequency spectrum.

19. The method that improves speech quality of a speech segment of claim 17 further comprising disabling the act of estimating the background noise when a transient noise is detected.

20. The method that improves speech quality of a speech segment of claim 17 further comprising converting the sound segment into the power domain.

21. The method that improves speech quality of a speech segment of claim 17 where the level of variable attenuation is based on a plurality of modeled line coordinate intercepts.

22. A non-transitory computer readable medium that retains software that improves a speech quality by modeling a background noise comprising:

a computer readable medium that retains a signal estimation logic, a modeling logic, and an attenuation logic that is accessible to and configured to be processed by a processor, where

the signal estimation logic determines the signal power of a desired signal within an input signal;

the modeling logic represents a plurality of background noises detected from the input signal through a plurality of substantially linear models, where the modeling logic fits a first line of the plurality of substantially linear models to a first frequency portion of the input signal, and fits a second line of the plurality of substantially linear models to a second frequency portion of the input signal; and

the attenuation logic approximates the level of suppression to be applied to the input signal in response to an output of the modeling logic, where the attenuation logic calculates a difference in slope between the first line and the second line, and calculates the level of suppression based on the difference in slope.

23. The computer readable medium of claim 22 further comprising a memory programmed to retain the plurality of substantially linear models.