Microphone array system

Abstract

A method and system for enhancing a target sound signal from multiple sound signals is provided. An array of an arbitrary number of sound sensors positioned in an arbitrary configuration receives the sound signals from multiple disparate sources. The sound signals comprise the target sound signal from a target sound source, and ambient noise signals. A sound source localization unit, an adaptive beamforming unit, and a noise reduction unit are in operative communication with the array of sound sensors. The sound source localization unit estimates a spatial location of the target sound signal from the received sound signals. The adaptive beamforming unit performs adaptive beamforming by steering a directivity pattern of the array of sound sensors in a direction of the spatial location of the target sound signal, thereby enhancing the target sound signal and partially suppressing the ambient noise signals, which are further suppressed by the noise reduction unit.

We claim: 
     
       1. A method for enhancing a target sound signal from a plurality of sound signals, comprising:
 providing a microphone array system comprising an array of sound sensors positioned in an arbitrary configuration, a sound source localization unit, an adaptive beamforming unit, and a noise reduction unit, wherein said sound source localization unit, said adaptive beamforming unit, and said noise reduction unit are in operative communication with said array of said sound sensors;   receiving said sound signals from a plurality of disparate sound sources by said sound sensors, wherein said received sound signals comprise said target sound signal from a target sound source among said disparate sound sources, and ambient noise signals;   determining a delay between each of said sound sensors and an origin of said array of said sound sensors as a function of distance between each of said sound sensors and said origin, a predefined angle between each of said sound sensors and a reference axis, and an azimuth angle between said reference axis and said target sound signal, when said target sound source that emits said target sound signal is in a two dimensional plane, wherein said delay is represented in terms of number of samples, and wherein said determination of said delay enables beamforming for arbitrary numbers of said sound sensors and a plurality of arbitrary configurations of said array of said sound sensors;   estimating a spatial location of said target sound signal from said received sound signals by said sound source localization unit;   performing adaptive beamforming for steering a directivity pattern of said array of said sound sensors in a direction of said spatial location of said target sound signal by said adaptive beamforming unit, wherein said adaptive beamforming unit enhances said target sound signal and partially suppresses said ambient noise signals; and   suppressing said ambient noise signals by said noise reduction unit for further enhancing said target sound signal.   
     
     
       2. The method of  claim 1 , wherein said spatial location of said target sound signal from said target sound source is estimated using a steered response power-phase transform by said sound source localization unit. 
     
     
       3. The method of  claim 1 , wherein said adaptive beamforming comprises:
 providing a fixed beamformer, a blocking matrix, and an adaptive filter in said adaptive beamforming unit;   steering said directivity pattern of said array of said sound sensors in said direction of said spatial location of said target sound signal from said target sound source by said fixed beamformer for enhancing said target sound signal, when said target sound source is in motion;   feeding said ambient noise signals to said adaptive filter by blocking said target sound signal received from said target sound source using said blocking matrix; and   adaptively filtering said ambient noise signals by said adaptive filter in response to detecting one of presence and absence of said target sound signal in said sound signals received from said disparate sound sources.   
     
     
       4. The method of  claim 3 , wherein said fixed beamformer performs fixed beamforming by filtering and summing output sound signals from said sound sensors. 
     
     
       5. The method of  claim 3 , wherein said adaptive filtering comprises sub-band adaptive filtering performed by said adaptive filter, wherein said sub-band adaptive filtering comprises:
 providing an analysis filter bank, an adaptive filter matrix, and a synthesis filter bank in said adaptive filter;   splitting said enhanced target sound signal from said fixed beamformer and said ambient noise signals from said blocking matrix into a plurality of frequency sub-bands by said analysis filter bank;   adaptively filtering said ambient noise signals in each of said frequency sub-bands by said adaptive filter matrix in response to detecting one of presence and absence of said target sound signal in said sound signals received from said disparate sound sources; and   synthesizing a full-band sound signal using said frequency sub-bands of said enhanced target sound signal by said synthesis filter bank.   
     
     
       6. The method of  claim 3 , wherein said adaptive beamforming further comprises detecting said presence of said target sound signal by an adaptation control unit provided in said adaptive beamforming unit and adjusting a step size for said adaptive filtering in response to detecting one of said presence and said absence of said target sound signal in said sound signals received from said disparate sound sources. 
     
     
       7. The method of  claim 1 , wherein said noise reduction unit performs noise reduction by using one of a Wiener-filter based noise reduction algorithm, a spectral subtraction noise reduction algorithm, an auditory transform based noise reduction algorithm, and a model based noise reduction algorithm. 
     
     
       8. The method of  claim 1 , wherein said noise reduction unit performs noise reduction in a plurality of frequency sub-bands, wherein said frequency sub-bands are employed by an analysis filter bank of said adaptive beamforming unit for sub-band adaptive beamforming. 
     
     
       9. A system for enhancing a target sound signal from a plurality of sound signals, comprising:
 an array of sound sensors positioned in an arbitrary configuration, wherein said sound sensors receive said sound signals from a plurality of disparate sound sources, wherein said received sound signals comprise said target sound signal from a target sound source among said disparate sound sources, and ambient noise signals;   a sound source localization unit that estimates a spatial location of said target sound signal from said received sound signals, by determining a delay between each of said sound sensors and an origin of said array of said sound sensors as a function of distance between each of said sound sensors and said origin, a predefined angle between each of said sound sensors and a reference axis, and an azimuth angle between said reference axis and said target sound signal, when said target sound source that emits said target sound signal is in a two dimensional plane, wherein said delay is represented in terms of number of samples, and wherein said determination of said delay enables beamforming for arbitrary numbers of said sound sensors and a plurality of arbitrary configurations of said array of said sound sensors;   an adaptive beamforming unit that steers directivity pattern of said array of said sound sensors in a direction of said spatial location of said target sound signal, wherein said adaptive beamforming unit enhances said target sound signal and partially suppresses said ambient noise signals; and   a noise reduction unit that suppresses said ambient noise signals for further enhancing said target sound signal.   
     
     
       10. The system of  claim 9 , wherein said sound source localization unit estimates said spatial location of said target sound signal from said target sound source using a steered response power-phase transform. 
     
     
       11. The system of  claim 9 , wherein said adaptive beamforming unit comprises:
 a fixed beamformer that steers said directivity pattern of said array of said sound sensors in said direction of said spatial location of said target sound signal from said target sound source for enhancing said target sound signal, when said target sound source is in motion;   a blocking matrix that feeds said ambient noise signals to an adaptive filter by blocking said target sound signal received from said target sound source; and   said adaptive filter that adaptively filters said ambient noise signals in response to detecting one of presence and absence of said target sound signal in said sound signals received from said disparate sound sources.   
     
     
       12. The system of  claim 11 , wherein said fixed beamformer performs fixed beamforming by filtering and summing output sound signals from said sound sensors. 
     
     
       13. The system of  claim 11 , wherein said adaptive filter comprises a set of sub-band adaptive filters comprising:
 an analysis filter bank that splits said enhanced target sound signal from said fixed beamformer and said ambient noise signals from said blocking matrix into a plurality of frequency sub-bands;   an adaptive filter matrix that adaptively filters said ambient noise signals in each of said frequency sub-bands in response to detecting one of presence and absence of said target sound signal in said sound signals received from said disparate sound sources; and   a synthesis filter bank that synthesizes a full-band sound signal using said frequency sub-bands of said enhanced target sound signal.   
     
     
       14. The system of  claim 9 , wherein said adaptive beamforming unit further comprises an adaptation control unit that detects said presence of said target sound signal and adjusts a step size for said adaptive filtering in response to detecting one of said presence and said absence of said target sound signal in said sound signals received from said disparate sound sources. 
     
     
       15. The system of  claim 9 , wherein said noise reduction unit is one of a Wiener-filter based noise reduction unit, a spectral subtraction noise reduction unit, an auditory transform based noise reduction unit, and a model based noise reduction unit. 
     
     
       16. The system of  claim 9 , further comprising one or more audio codecs that convert said sound signals in an analog form of said sound signals into digital sound signals and reconverts said digital sound signals into said analog form of said sound signals. 
     
     
       17. The system of  claim 9 , wherein said noise reduction unit performs noise reduction in a plurality of frequency sub-bands employed by an analysis filter bank of said adaptive beamforming unit for sub-band adaptive beamforming. 
     
     
       18. The system of  claim 9 , wherein said array of said sound sensors is one of a linear array of said sound sensors, a circular array of said sound sensors, and an arbitrarily distributed coplanar array of said sound sensors. 
     
     
       19. The method of  claim 1 , wherein said delay (τ) is determined by a formula τ=f s *t, wherein f s  is a sampling frequency and t is a time delay. 
     
     
       20. A method for enhancing a target sound signal from a plurality of sound signals, comprising:
 providing a microphone array system comprising an array of sound sensors positioned in an arbitrary configuration, a sound source localization unit, an adaptive beamforming unit, and a noise reduction unit, wherein said sound source localization unit, said adaptive beamforming unit, and said noise reduction unit are in operative communication with said array of said sound sensors;   receiving said sound signals from a plurality of disparate sound sources by said sound sensors, wherein said received sound signals comprise said target sound signal from a target sound source among said disparate sound sources, and ambient noise signals;   determining a delay between each of said sound sensors and an origin of said array of said sound sensors as a function of distance between each of said sound sensors and said origin, a predefined angle between each of said sound sensors and a first reference axis, an elevation angle between a second reference axis and said target sound signal, and an azimuth angle between said first reference axis and said target sound signal, when said target sound source that emits said target sound signal is in a three dimensional plane, wherein said delay is represented in terms of number of samples, and wherein said determination of said delay enables beamforming for arbitrary numbers of said sound sensors and a plurality of arbitrary configurations of said array of said sound sensors;   estimating a spatial location of said target sound signal from said received sound signals by said sound source localization unit;   performing adaptive beamforming for steering a directivity pattern of said array of said sound sensors in a direction of said spatial location of said target sound signal by said adaptive beamforming unit, wherein said adaptive beamforming unit enhances said target sound signal and partially suppresses said ambient noise signals; and   suppressing said ambient noise signals by said noise reduction unit for further enhancing said target sound signal.   
     
     
       21. A system for enhancing a target sound signal from a plurality of sound signals, comprising:
 an array of sound sensors positioned in an arbitrary configuration, wherein said sound sensors receive said sound signals from a plurality of disparate sound sources, wherein said received sound signals comprise said target sound signal from a target sound source among said disparate sound sources, and ambient noise signals;   a sound source localization unit that estimates a spatial location of said target sound signal from said received sound signals as a function of distance between each of said sound sensors and said origin, a predefined angle between each of said sound sensors and a first reference axis, an elevation angle between a second reference axis and said target sound signal, and an azimuth angle between said first reference axis and said target sound signal, when said target sound source that emits said target sound signal is in a three dimensional plane, wherein said delay is represented in terms of number of samples, and wherein said determination of said delay enables beamforming for arbitrary numbers of said sound sensors and a plurality of arbitrary configurations of said array of said sound sensors;   an adaptive beamforming unit that steers directivity pattern of said array of said sound sensors in a direction of said spatial location of said target sound signal, wherein said adaptive beamforming unit enhances said target sound signal and partially suppresses said ambient noise signals; and   a noise reduction unit that suppresses said ambient noise signals for further enhancing said target sound signal.   
     
     
       22. A method for enhancing a target sound signal from a plurality of sound signals, comprising:
 providing a microphone array system comprising an array of sound sensors positioned in a linear, circular, or other configuration, a sound source localization unit, an adaptive beamforming unit, a noise reduction unit, and an echo cancellation unit, wherein said sound source localization unit, said adaptive beamforming unit, said noise reduction unit, and said echo cancellation unit are implemented in a digital signal processor, and wherein said digital signal processor is in operative communication with said array of said sound sensors;   receiving said sound signals from a plurality of disparate sound sources by said sound sensors, wherein said received sound signals comprise said target sound signal from a target sound source among said disparate sound sources, and ambient noise signals;   determining a delay between each of said sound sensors and an origin of said array of said sound sensors as a function of distance between each of said sound sensors and said origin, a predefined angle between each of said sound sensors and a reference axis, and an azimuth angle between said reference axis and said target sound signal, when said target sound source that emits said target sound signal is in a two dimensional plane, wherein said delay is represented in terms of number of samples, and wherein said determination of said delay enables beamforming for said array of said sound sensors in a plurality of configurations;   estimating a location of said target sound signal from said received sound signals by said sound source localization unit;   performing adaptive beamforming for steering a directivity pattern of said array of said sound sensors in a direction of said location of said target sound signal by said adaptive beamforming unit, wherein said adaptive beamforming unit enhances said target sound signal and partially suppresses said ambient noise signals;   performing echo cancellation by said echo cancellation unit for further enhancing said target sound signal; and   suppressing said ambient noise signals by said noise reduction unit for further enhancing said target sound signal.   
     
     
       23. The method of claim 22, wherein said location of said target sound signal from said target sound source is estimated using a steered response power-phase transform by said sound source localization unit. 
     
     
       24. The method of claim 22, wherein said adaptive beamforming comprises:
 providing a fixed beamformer, a blocking matrix, and an adaptive filter in said adaptive beamforming unit;   steering said directivity pattern of said array of said sound sensors in said direction of said location of said target sound signal from said target sound source by said fixed beamformer for enhancing said target sound signal, when said target sound source is in motion;   feeding said ambient noise signals to said adaptive filter by blocking said target sound signal received from said target sound source using said blocking matrix; and   adaptively filtering said ambient noise signals by said adaptive filter in response to voice activity detection, wherein said voice activity detection comprises detecting one of presence and absence of said target sound signal in said sound signals received from said disparate sound sources.   
     
     
       25. The method of claim 24, wherein said fixed beamformer performs fixed beamforming by one of filtering and summing output sound signals from said sound sensors, and delaying and summing output sound signals from said sound sensors. 
     
     
       26. The method of claim 24, wherein said adaptive filtering comprises sub-band adaptive filtering performed by said adaptive filter, and wherein said sub-band adaptive filtering comprises:
 providing an analysis filter bank, an adaptive filter matrix, and a synthesis filter bank in said adaptive filter;   splitting said enhanced target sound signal from said fixed beamformer and said ambient noise signals from said blocking matrix into a plurality of frequency sub-bands by said analysis filter bank;   adaptively filtering said ambient noise signals in each of said frequency sub-bands by said adaptive filter matrix in response to said detection of one of said presence and said absence of said target sound signal in said sound signals received from said disparate sound sources; and   synthesizing a full-band sound signal using said frequency sub-bands of said enhanced target sound signal by said synthesis filter bank.   
     
     
       27. The method of claim 24, wherein said adaptive beamforming further comprises detecting said presence of said target sound signal by an adaptation control unit provided in said adaptive beamforming unit and adjusting a step size for said adaptive filtering in response to said detection of one of said presence and said absence of said target sound signal in said sound signals received from said disparate sound sources. 
     
     
       28. The method of claim 22, wherein said noise reduction unit performs noise reduction by using one of a Wiener-filter based noise reduction algorithm, a spectral subtraction noise reduction algorithm, an auditory transform based noise reduction algorithm, and a model based noise reduction algorithm. 
     
     
       29. The method of claim 22, wherein said noise reduction unit performs noise reduction in a plurality of frequency sub-bands, wherein said frequency sub-bands are employed by an analysis filter bank of said adaptive beamforming unit for sub-band adaptive beamforming, wherein said sound source localization unit calculates said delay (τ) based on said number of samples within a time period and a time delay for said target sound signal to travel said distance between each of said sound sensors in said microphone array and said origin of said array of said sound sensors, and wherein said distance between said each of said sound sensors in the microphone array and said origin of said array of said sound sensors is either a same distance or a different distance. 
     
     
       30. A microphone array system for enhancing a target sound signal from a plurality of sound signals, comprising:
 an array of sound sensors positioned in a linear, circular, or other configuration, wherein said sound sensors receive said sound signals from a plurality of disparate sound sources, wherein said received sound signals comprise said target sound signal from a target sound source among said disparate sound sources, and ambient noise signals;   a digital signal processor, said digital signal processor comprising:
 a sound source localization unit that estimates a location of said target sound signal from said received sound signals, by determining a delay between each of said sound sensors and an origin of said array of said sound sensors as a function of distance between each of said sound sensors and said origin, a predefined angle between each of said sound sensors and a reference axis, and an azimuth angle between said reference axis and said target sound signal, when said target sound source that emits said target sound signal is in a two dimensional plane, wherein said delay is represented in terms of number of samples, and wherein said determination of said delay enables beamforming for said array of sound sensors in a plurality of configurations; 
 an adaptive beamforming unit that steers directivity pattern of said array of said sound sensors in a direction of said location of said target sound signal, wherein said adaptive beamforming unit enhances said target sound signal and partially suppresses said ambient noise signals; 
 an echo cancellation unit that performs echo cancellation for further enhancing said target sound signal; and 
 a noise reduction unit that suppresses said ambient noise signals for further enhancing said target sound signal. 
   
     
     
       31. The system of claim 30, wherein said sound source localization unit estimates said location of said target sound signal from said target sound source using a steered response power-phase transform. 
     
     
       32. The system of claim 30, wherein said adaptive beamforming unit comprises:
 a fixed beamformer that steers said directivity pattern of said array of said sound sensors in said direction of said location of said target sound signal from said target sound source for enhancing said target sound signal, when said target sound source is in motion;   a blocking matrix that feeds said ambient noise signals to an adaptive filter by blocking said target sound signal received from said target sound source; and   said adaptive filter adaptively filters said ambient noise signals in response to voice activity detection, wherein said voice activity detection comprises detecting one of presence and absence of said target sound signal in said sound signals received from said disparate sound sources.   
     
     
       33. The system of claim 32, wherein said fixed beamformer performs fixed beamforming by one of filtering and summing output sound signals from said sound sensors, and delaying and summing output sound signals from said sound sensors. 
     
     
       34. The system of claim 32, wherein said adaptive filter comprises a set of sub-band adaptive filters comprising:
 an analysis filter bank that splits said enhanced target sound signal from said fixed beamformer and said ambient noise signals from said blocking matrix into a plurality of frequency sub-bands;   an adaptive filter matrix that adaptively filters said ambient noise signals in each of said frequency sub-bands in response to said detection of one of said presence and said absence of said target sound signal in said sound signals received from said disparate sound sources; and   a synthesis filter bank that synthesizes a full-band sound signal using said frequency sub-bands of said enhanced target sound signal.   
     
     
       35. The system of claim 32, wherein said adaptive beamforming unit further comprises an adaptation control unit that detects said presence of said target sound signal and adjusts a step size for said adaptive filtering in response to said detection of one of said presence and said absence of said target sound signal in said sound signals received from said disparate sound sources. 
     
     
       36. The system of claim 30, wherein said noise reduction unit is one of a Wiener-filter based noise reduction unit, a spectral subtraction noise reduction unit, an auditory transform based noise reduction unit, and a model based noise reduction unit, wherein said noise reduction unit performs noise reduction in a plurality of frequency sub-bands employed by an analysis filter bank of said adaptive beamforming unit for sub-band adaptive beamforming, wherein said sound source localization unit calculates said delay (τ) based on said number of samples within a time period and a time delay for said target sound signal to travel said distance between each of said sound sensors in said microphone array and said origin of said array of said sound sensors, and wherein said distance between said each of said sound sensors in the microphone array and said origin of said array of said sound sensors is either a same distance or a different distance. 
     
     
       37. The system of claim 30, further comprising one or more audio codecs that convert said sound signals in an analog form of said sound signals into digital sound signals and reconverts said digital sound signals into said analog form of said sound signals. 
     
     
       38. A microphone array system for enhancing a target sound signal from a plurality of sound signals, comprising:
 an array of sound sensors, wherein said sound sensors receive said sound signals from a plurality of disparate sound sources, wherein said received sound signals comprise said target sound signal from a target sound source among said disparate sound sources, and ambient noise signals;   a digital signal processor, said digital signal processor comprising:
 a sound source localization unit that estimates a location of said target sound signal from said received sound signals by determining a delay between each of said sound sensors and a reference point of said array of said sound sensors as a function of distance between each of said sound sensors and said reference point and an angle of each of said sound sensors biased from a reference axis; 
 a beamforming unit that enhances said target sound signal and partially suppresses said ambient noise signals; 
 an echo cancellation unit that performs echo cancellation and further enhances said target sound signal; and 
 a noise reduction unit that suppresses said ambient noise signals and further enhances said target sound signal. 
   
     
     
       39. The system of claim 38, wherein said microphone array system is implemented in one of devices with speech acquisition capability, hands-free devices, handheld devices, conference phones and video conferencing applications, wherein said handheld devices comprise smart phones, tablet computers and laptop computers, and wherein said array of said sound sensors is one of a linear array of said sound sensors, a circular array of said sound sensors, and other types of array of said sound sensors. 
     
     
       40. The method of claim 22, wherein said microphone array system is implemented in one of devices with speech acquisition capability, hands-free devices, handheld devices, conference phones and video conferencing applications, wherein said handheld devices comprise smart phones, tablet computers and laptop computers. 
     
     
       41. The system of claim 30, wherein said microphone array system is implemented in one of devices with speech acquisition capability, hands-free devices, handheld devices, conference phones and video conferencing applications, wherein said handheld devices comprise smart phones, tablet computers and laptop computers.