Dual Adaptive Structure for Speech Enhancement
AssigneeAlon Konchitsky
InventorsAlon Konchitsky, Alberto D. Berstein, Hariharan Ganapathy Kathirvelu, Sandeep Kulakcherla, William Ribble
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Abstract
A clear, high quality voice signal with a high signal-to-noise ratio is achieved by use of an adaptive noise reduction scheme with two microphones in close proximity. The method includes the use of two omini directional microphones in a highly directional mode, and then applying an adaptive noise cancellation algorithm to reduce the noise.
Claims6
A method of improving the signal to noise ratio in a communication system, the method comprising:
- a) acquiring one or more buffers of sound samples from a rear microphone and a front microphone, resulting in a rear microphone signal and a front microphone signal;
- b) applying a propagation delay between the two microphones for a length of time equal to one sample, resulting in a delayed rear microphone signal and a delayed front microphone signal;
- c) subtracting the delayed rear microphone signal from the rear microphone signal;
- d) subtracting the delayed front microphone signal from the front microphone signal;
- e) using a first adaptive filter, the first filter calculating weights adaptively, as the ratios of the cross-correlation between the two microphones Rxy, and the auto-correlation of the rear microphone, Ryy, and averaging the auto-correlation and cross-correlation for smoothing purposes;
- f) subtracting the output of the first adaptive filter from a signal obtained by subtracting the delayed rear microphone signal from the front microphone signal, giving a first level of output processing;
- g) using a voice activity detector to determine speech and non-speech regions and to control the first adaptive filter and a second adaptive filter;
- h) during non-speech regions, the voice activity detector is in an off position and weights of the second adaptive filter are updated, and the second adaptive filter receives a signal obtained by subtracting the rear microphone signal from the delayed front microphone signal, the output from the second adaptive filter is sent to the and a second level processing unit;
- i) during speech regions, the voice activity detector is in an on position and freezes adaptive weight calculations and sends the resulting output to the second level processing unit;
- and
- j) the second level processing unit removes residual noise left over from the first processing level.
The method of claim 2 wherein the averaging of the auto-correlation and cross-correlation is achieved by the following equation:
- W
- opt
- =
- R
- xy
- R
- yy
- R
- xy
- =
- α
-
-
- R
- xy_prev
- +
- (
- 1
- -
- α
- )
-
- R
- xy
- R
- yy
- =
- α
-
-
- R
- yy_prev
- +
- (
- 1
- -
- α
- )
-
- R
- yy
- and the value of a can be chosen to be in the range 0.75 to 0.95.
An adaptive directionality microphone system, the system comprising:
- a) a back microphone sends input into a delay element wherein the back microphone signal delayed by a unit of time t;
- b) a cardioid x(n) component subtracts output of a rear microphone signal from the output of the delay element to give cardioid signal, y(n), with a null at 0°;
- c) cardioids signal y(n) is filtered using a first adaptive filter W1(z) which generates adaptive weights, to give an output a(n);
- d) a subtraction component subtracts the output of the first adaptive filter from x(n) to give a directional signal, z(n)
Dependencies are inferred from the claim text.