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US 7,817,808 B2Granted
Dual adaptive structure for speech enhancement
OwnerAlon Konchitsky, Alberto D Berstein +3inventors
InventorsAlon Konchitsky, Alberto D Berstein, Hariharan Ganapathy Kathirvelu, Sandeep Kulakcherla, William Martin 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.
Claims3
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 back microphone and a front microphone, resulting in a back 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 back microphone signal and a delayed front microphone signal;
- c) subtracting the delayed back microphone signal from the front microphone signal;
- d) subtracting the back microphone signal from the delayed front microphone signal;
- e) using a first adaptive filter, the first adaptive filter calculating weights adaptively, as the ratios of the cross-correlation between the two microphones Rxy, and the auto-correlation of the back 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 back 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 back microphone signal from the delayed front microphone signal, the output from the second adaptive filter is sent to a second level processing unit;
- i) during speech regions, the voice activity detector in is an on position and freezes adaptive weight calculations and send 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.
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A method comprising:
- a) directing a front microphone input into a delay element wherein the front microphone signal is delayed by a unit of time t and a back microphone input into a delay element wherein the back microphone signal is delayed by a unit of time t;
- b) obtaining a cardioid x(n) signal obtained by subtracting the output of the delayed back microphone signal from the front microphone signal and a cardioid signal, y(n), obtained by subtracting the back microphone signal from the delayed front microphone signal;
- c) filtering cardioids signal y(n) by using a first adaptive filter W1(z) which generates adaptive weights, to give an output a(n);
- d) subtracting, by use of a subtraction component that subtracts the output of the first adaptive filter from x(n) to give a directional signal, z(n);
- e) the filter coefficients are adaptively estimated to minimize the power of interfering noise;
- f) the polar pattern of the system output z(n) is a combination of x(n) and y(n) and determined by the filter W1(z);
- g) combining an adaptive noise cancellation method, the adaptive noise cancellation method comprising:
- i) causing the signal from the back microphone to be delayed by a time period one sample and the resulting signal is subtracted from the front microphone signal to produce a cardioid, x(n) with a null at 180°;
- ii) causing the signal from the front microphone to be delayed by a time period of one sample, to produce a delayed front microphone signal, the back microphone signal is subtracted from the delayed front microphone signal to produce a cardioid, y(n) with a null at 0°;
- iii) filtering the signal y(n) by using a first adaptive filter W1(z) to give an output a(n);
- iv) subtracting the output of the first adaptive filter from the signal x(n) to produce directional signal z(n),
- v) using signal v(n) as a reference input to a second adaptive filter W2(z);
- vi) detecting speech and non-speech regions of directional signal z(n) by use of a voice activity detector, detecting speech and giving the signal as the primary input to the second adaptive filter which in turn produces an output similar to the noise that remains in the z(n) signal;
- and
- vii) subtracting the output of the second adaptive filter from the directional signal z(n).
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Dependencies are inferred from the claim text.