Ultra small microphone array
First Claim
1. A method for digitally processing a signal from an array of two or more microphones M0 . . . MM, the method comprising:
- producing a discrete time domain input signal xm(t) at a runtime from each of the two or more microphones M0 . . . MM, where M is greater than or equal to 1;
determining a listening direction of the microphone array with a digital signal processing system having a digital processor coupled to a memory byforming analysis frames of a pre-recorded signal stored in the memory from a source located in a preferred known listening direction with respect to the microphone array for a predetermined period of time at predetermined intervals using the processor,transforming the analysis frames into the frequency domain using the processor,estimating a calibration covariance matrix from vectors formed from the analysis frames that have been transformed into the frequency domain using the processor,computing an eigenmatrix of the calibration covariance matrix, andcomputing an inverse of the eigenmatrix;
using the known listening direction in a semi-blind source separation implemented by the processor to select a set of N finite impulse response filter coefficients bi, where N is a positive integer.
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Abstract
Methods and apparatus for signal processing are disclosed. A discrete time domain input signal xm(t) may be produced from an array of microphones M0 . . . MM. A listening direction may be determined for the microphone array. The listening direction is used in a semi-blind source separation to select the finite impulse response filter coefficients b0, b1 . . . , bN to separate out different sound sources from input signal xm(t). One or more fractional delays may optionally be applied to selected input signals xm(t) other than an input signal x0(t) from a reference microphone M0. Each fractional delay may be selected to optimize a signal to noise ratio of a discrete time domain output signal y(t) from the microphone array. The fractional delays may be selected to such that a signal from the reference microphone M0 is first in time relative to signals from the other microphone(s) of the array. A fractional time delay Δ may optionally be introduced into an output signal y(t) so that: y(t+Δ)=x(t+Δ)*b0+x(t−1+Δ)*b1+x(t−2+Δ)*b2+ . . . +x(t−N+Δ)bN, where Δ is between zero and ±1.
143 Citations
29 Claims
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1. A method for digitally processing a signal from an array of two or more microphones M0 . . . MM, the method comprising:
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producing a discrete time domain input signal xm(t) at a runtime from each of the two or more microphones M0 . . . MM, where M is greater than or equal to 1; determining a listening direction of the microphone array with a digital signal processing system having a digital processor coupled to a memory by forming analysis frames of a pre-recorded signal stored in the memory from a source located in a preferred known listening direction with respect to the microphone array for a predetermined period of time at predetermined intervals using the processor, transforming the analysis frames into the frequency domain using the processor, estimating a calibration covariance matrix from vectors formed from the analysis frames that have been transformed into the frequency domain using the processor, computing an eigenmatrix of the calibration covariance matrix, and computing an inverse of the eigenmatrix; using the known listening direction in a semi-blind source separation implemented by the processor to select a set of N finite impulse response filter coefficients bi, where N is a positive integer. - View Dependent Claims (2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15)
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16. A signal processing apparatus, comprising:
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an array of two or more microphones M0 . . . MM wherein each of the two or more microphones is adapted to produce a discrete time domain input signal xm(t) at a runtime; one or more processors coupled to the array of two or more microphones; and a memory coupled to the array of two or more microphones and the processor, the memory having embodied therein a set of processor readable instructions configured to implement a method for digitally processing a signal, the processor readable instructions including; one or more instructions for determining a listening direction of the microphone array from the discrete time domain input signals xm(t) by forming analysis frames of a pre-recorded a signal from a source located in a preferred known listening direction with respect to the microphone array for a predetermined period of time at predetermined intervals, transforming the analysis frames into the frequency domain, estimating a calibration covariance matrix from vectors formed from the analysis frames that have been transformed into the frequency domain, computing an eigenmatrix of the calibration covariance matrix, and computing an inverse of the eigenmatrix; and one or more instructions for using the known listening direction in a semi-blind source separation to select filtering functions to separate out two or more sources of sound from the discrete time domain input signals xm(t). - View Dependent Claims (17, 18, 19, 20, 21, 22, 23, 24, 25, 26)
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27. A method for digitally processing a signal from an array of two or more microphones M0 . . . MM, the method comprising:
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receiving an audio signal at each of the two or more microphones M0 . . . MM; producing a discrete time domain input signal xm(t) at a runtime from each of the two or more microphones M0 . . . MM; determining a listening direction of the microphone array with a digital signal processing system having a digital processor by forming analysis frames of a pre-recorded a signal from a source located in a preferred known listening direction with respect to the microphone array for a predetermined period of time at predetermined intervals using the processor, transforming the analysis frames into the frequency domain using the processor, estimating a calibration covariance matrix from vectors formed from the analysis frames that have been transformed into the frequency domain using the processor, computing an eigenmatrix of the calibration covariance matrix using the processor, and computing an inverse of the eigenmatrix using the processor applying one or more fractional delays to one or more of the time domain input signals xm(t) other than an input signal x0(t) from a reference microphone M0 using the processor, wherein each fractional delay is selected to optimize a signal to noise ratio of an output signal from the microphone array and wherein the fractional delays are selected to such that a signal from the reference microphone M0 is first in time relative to signals from the other microphone(s) of the array. - View Dependent Claims (28, 29)
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Specification