Acoustic keystroke transient canceler for speech communication terminals using a semi-blind adaptive filter model

Patent metadata

Field	Value
Publication number	US-9881630-B2
Application number	US-201514984373-A
Country	US
Kind code	B2
Filing date	Dec 30, 2015
Priority date	Dec 30, 2015
Publication date	Jan 30, 2018
Grant date	Jan 30, 2018

Abstract

Provided are methods and systems for acoustic keystroke transient cancellation/suppression for user communication devices using a semi-blind adaptive filter model. The methods and systems are designed to overcome existing problems in transient noise suppression by taking into account some less-defective signal as side information on the transients and also accounting for acoustic signal propagation, including the reverberation effects, using dynamic models. The methods and systems take advantage of a synchronous reference microphone embedded in the keyboard of the user device, and utilize an adaptive filtering approach exploiting the knowledge of this keybed microphone signal.

First claim

The invention claimed is:

1. A system for suppressing transient noise, the system comprising:
a plurality of input sensors that input audio signals captured from one or more sources, wherein the audio signals contain voice data and transient noise captured by the input sensors;
a reference sensor that inputs a reference signal containing data about the transient noise, wherein the reference sensor is located separately from the input sensors;
a semi-blind adaptive single-input and multi-output (SIMO) filtering structure that includes:
a plurality of filters that selectively filter the transient noise from the audio signals to extract the voice data based on the data contained in the reference signal, and output an audio signal containing the extracted voice data; and
a single-channel equalizing post-filter that filters linear distortion from the audio signal containing the extracted voice data and that outputs an enhanced audio single containing the extracted voice data, wherein the single-channel equalizing post-filter includes a filter that is an inversion of one of the plurality of filters.

2. The system of claim 1 , wherein each of the plurality of filters is a broadband finite impulse response filter.

3. The system of claim 1 , wherein the plurality of filters include:
an adaptive foreground filter; and
an adaptive background filter, wherein
the foreground filter adaptively filters the transient noise to produce the output audio signal, and
the background filter controls the adaptation of the foreground filter.

4. The system of claim 3 , wherein the background filter controls the adaptation of the foreground filter based on the data contained in the reference signal.

5. The system of claim 3 , wherein the background filter controls the adaptation of the foreground filter in response to transient noise being detected in the audio signals.

6. The system of claim 3 , wherein the background filter controls the adaptation of the foreground filter based on one or more of a power of the reference signal, a ratio of a linear approximation to a nonlinearity contribution of the reference signal, and spatio-temporal source signal activity data associated with the reference signal.

7. The system of claim 3 , wherein the background filter controls the adaptation of the foreground filter based on a power of the reference signal, a ratio of a linear approximation to the nonlinearity contribution of the reference signal, and spatio-temporal source signal activity data associated with the reference signal.

8. The system of claim 1 , wherein the transient noise contained in the audio signals is a keystroke noise generated from a keybed of a user device.

9. The system of claim 1 , wherein the input sensors and the reference sensor are microphones.

10. The system of claim 1 , wherein the plurality of filters filter the transient noise from the audio signals by subtracting the reference signal input from the reference sensor.

11. A method for suppressing transient noise, the method comprising:
receiving, from a plurality of input sensors, input audio signals captured from one or more sources, wherein the audio signals contain voice data and transient noise captured by the input sensors;
receiving, from a reference sensor, a reference signal containing data about the transient noise, wherein the reference sensor is located separately from the input sensors;
selectively filtering the transient noise from the audio signals to extract the voice data based on the data contained in the reference signal;
outputting an audio signal containing the extracted voice data;
filtering, by a single-channel equalizing post filter, linear distortion from the audio signal containing the extracted voice data, wherein the single-channel equalizing post-filter includes a filter that is an inversion of the plurality of filters; and
outputting an enhanced audio single containing the extracted voice data.

12. The method of claim 11 , wherein the transient noise is selectively filtered from the audio signals using broadband finite impulse response filters.

13. The method of claim 11 , further comprising:
adapting a foreground filter to adaptively filter the transient noise to produce the output audio signal.

14. The method of claim 13 , further comprising:
controlling the adaptation of the foreground filter using a background filter.

15. The method of claim 14 , wherein the background filter controls the adaptation of the foreground filter based on the data contained in the reference signal.

16. The method of claim 14 , wherein the background filter controls the adaptation of the foreground filter in response to transient noise being detected in the audio signals.

17. The method of claim 14 , wherein the background filter controls the adaptation of the foreground filter based on one or more of a power of the reference signal, a ratio of a linear approximation to a nonlinearity contribution of the reference signal, and spatio-temporal source signal activity data associated with the reference signal.

18. The method of claim 14 , wherein the background filter controls the adaptation of the foreground filter based on a power of the reference signal, a ratio of a linear approximation to the nonlinearity contribution of the reference signal, and spatio-temporal source signal activity data associated with the reference signal.

19. The method of claim 11 , wherein the transient noise contained in the audio signals is a keystroke noise generated from a keybed of a user device.

20. The method of claim 11 , wherein the input sensors and the reference sensor are microphones.

Assignees

• Google Inc
• Google Llc

Inventors

• Buchner Herbert
• Godsill Simon J
• Skoglund Jan

Classifications

• G10L21/0224

  Processing in the time domain · CPC title

• G10L21/0208

  Noise filtering · CPC title

• G10L2021/02165

  Two microphones, one receiving mainly the noise signal and the other one mainly the speech signal · CPC title

• G10L21/0232

  Processing in the frequency domain · CPC title

• G10L2021/02161

  Number of inputs available containing the signal or the noise to be suppressed · CPC title