IP Library Granted Patent US 11,741,982
Granted Patent B2
US 11,741,982 · App. 17/494,512 · Granted Aug 29, 2023

Contextual beamforming to improve signal-to-noise ratio sensitive audio input processing efficiency in noisy environments

Inventors: Cola Hung Shih (Taipei, TW); Vivek Viswanathan Iyer (Austin, TX)
Assignee: Dell Products L.P.
G10L21/0216G10L17/00G10L25/30G10L25/78H04R3/005H04R3/04H04R5/04G10L2021/02082G10L2021/02166
View Patent ↗
Loading inventors, assignments & file history…
Monitor This Case
Get email alerts when status or documents change.
Order Certified Copies
Most orders are placed with the USPTO same day — all within 24 business hours.
Order via The Patent Place →
Pre-filled with this patent's details
Quick Facts
Patent No.
US 11,741,982
App. No.
17/494,512
Granted
Aug 29, 2023
Kind
B2
Abstract

An audio processing system includes a microphone array, a speech detection system, and a neural network noise reduction module. The microphone array includes at least two microphones and provides an audio signal from an environment surrounding the microphone array. The speech detection system receives the audio signal, and processes the audio signal to a) detect that a first user is speaking, b) determine a first direction relative to the audio array when the first user is located at a first location within the environment, and c) provide beamforming processing on the audio signal in the first direction, and to provide a processed audio signal based upon the beamforming processing. The neural network noise reduction module reduces noise in the processed audio signal.

Claims (28)

1. An audio processing system, comprising:

a microphone array including at least two microphones and configured to provide an audio signal from an environment surrounding the microphone array; and

a speech detection system configured to receive the audio signal, and to process the audio signal to a) characterize a first voice of a first user and a second voice of a second user, b) detect that the first user is speaking utilizing a speaker diarization model to distinguish the first voice from the second voice, c) determine a first direction relative to the audio array when the first user is located at a first location within the environment, d) provide beamforming processing on the audio signal in the first direction, and e) provide a processed audio signal based upon the beamforming processing

wherein the audio processing system is configured to reduce noise in the processed audio signal.

2. The audio processing system of claim 1 , wherein the speech detection system is further configured to determine that the first user has moved form the first location within the environment to a second location within the environment.

3. The audio processing system of claim 2 , wherein the speech detection system is further configured to determine a second direction relative to the audio array when the first user is located at the second location.

4. The audio processing system of claim 3 , wherein the speech detection system is further configured to provide beamforming processing on the audio signal in the second direction.

5. The audio processing system of claim 1 , wherein the speech detection system is further configured to process the audio signal to a) detect that the second user is speaking, b) determine a third direction relative to the audio array when the second user is located at a third location within the environment, and c) provide beamforming processing on the audio signal in the third direction.

6. The audio processing system of claim 1 , wherein, in determining the first direction, the speech detection system is further configured to determine a time delay in the audio signal between the microphones of the microphone array.

7. The information handling system of claim 1 , wherein, in detecting that the first user is speaking, the speech detection system utilizes a voice identification algorithm including at least one of a frequency estimation algorithm, a pattern matching algorithm, a neural network algorithm, a matrix representation algorithm, a vector quantization algorithm, and a decision tree algorithm.

8. A method comprising:

providing, by a microphone array of an audio processing system, an audio signal from an environment surrounding the microphone array, wherein the microphone array includes at least two microphones;

characterizing a first voice of a first user and a second voice of a second user;

processing the audio signal to a) detect that the first user is speaking utilizing a speaker diarization model to distinguish the first voice from the second voice, b) determine a first direction relative to the audio array when the first user is located at a first location within the environment, and c) provide beamforming processing on the audio signal in the first direction;

providing a processed audio signal based upon the beamforming processing; and

reducing noise in the processed audio signal.

9. The method of claim 8 , further comprising determining that the first user has moved form the first location within the environment to a second location within the environment.

10. The method of claim 9 , further comprising determining a second direction relative to the audio array when the first user is located at the second location.

11. The method of claim 10 , further comprising providing beamforming processing on the audio signal in the second direction.

12. The method of claim 8 , wherein detecting that the first user is speaking is based upon the characterization of the first voice.

13. The method of claim 8 , further comprising processing the audio signal to a) detect that the second user is speaking, b) determine a third direction relative to the audio array when the second user is located at a third location within the environment, and c) provide beamforming processing on the audio signal in the third direction.

14. The method of claim 8 wherein, in detecting that the first user is speaking, the method further comprises utilizing at least one of a frequency estimation algorithm, a pattern matching algorithm, a neural network algorithm, a matrix representation algorithm, a vector quantization algorithm, and a decision tree algorithm.

15. An information handling system, comprising:

a microphone array including at least two microphones and configured to provide an audio signal from an environment surrounding the microphone array; and

a processor configured to provide an audio processing system including:

a speech detection system configured to receive the audio signal, and to process the audio signal to a) characterize a first voice of a first user and a second voice of a second user, b) detect that the first user is speaking utilizing a speaker diarization model to distinguish the first voice from the second voice, c) determine a first direction relative to the audio array when the first user is located at a first location within the environment, d) provide beamforming processing on the audio signal in the first direction, and e) provide a processed audio signal based upon the beamforming processing; and

a neural network noise reduction module configured to reduce noise in the processed audio signal.

16. The information handling system of claim 15 , wherein, in detecting that the first user is speaking, the speech detection system utilizes a voice identification algorithm including at least one of a frequency estimation algorithm, a pattern matching algorithm, a neural network algorithm, a matrix representation algorithm, a vector quantization algorithm, and a decision tree algorithm.

Assignments (1)
ASSIGNMENT OF ASSIGNOR'S INTEREST Recorded Oct 5, 2021
From: SHIH, COLA HUNG; IYER, VIVEK VISWANATHAN
To: DELL PRODUCTS, LP
Reel/Frame 057706/0687 →
Continuity (1)
Related Publication 20230109066A1 · Apr 6, 2023