公开(公告)号：US10045137B2

公开(公告)日：2018-08-07

申请号：US15639263

申请日：2017-06-30

Applicant: Google Inc.

Inventor： Jan Skoglund ,  Yiteng Huang ,  Alejandro Luebs

IPC: H04R29/00 ,  H04B3/23 ,  G01N29/44 ,  G10L21/0208

Abstract: Techniques of performing acoustic echo cancellation involve providing a bi-magnitude filtering operation that performs a first filtering operation when a magnitude of an incoming audio signal to be output from a loudspeaker is less than a specified threshold and a second filtering operation when the magnitude of the incoming audio signal is greater than the threshold. The first filtering operation may take the form of a convolution between the incoming audio signal and a first impulse response function. The second filtering operation may take the form of a convolution between a nonlinear function of the incoming audio signal and a second impulse response function. For such a convolution, the bi-magnitude filtering operation involves providing, as the incoming audio signal, samples of the incoming audio signal over a specified window of time. The first and second impulse response functions may be determined from an input signal input into a microphone.

公开(公告)号：US20180007482A1

公开(公告)日：2018-01-04

申请号：US15639263

申请日：2017-06-30

Applicant: Google Inc.

Inventor： Jan Skoglund ,  Yiteng Huang ,  Alejandro Luebs

IPC: H04R29/00 ,  G01N29/44 ,  H04B3/23 ,  G10L21/0208

CPC classification number: H04R29/001 ,  G01N29/4427 ,  G10L2021/02082 ,  H04B3/237 ,  H04M9/082

Abstract: Techniques of performing acoustic echo cancellation involve providing a bi-magnitude filtering operation that performs a first filtering operation when a magnitude of an incoming audio signal to be output from a loudspeaker is less than a specified threshold and a second filtering operation when the magnitude of the incoming audio signal is greater than the threshold. The first filtering operation may take the form of a convolution between the incoming audio signal and a first impulse response function. The second filtering operation may take the form of a convolution between a nonlinear function of the incoming audio signal and a second impulse response function. For such a convolution, the bi-magnitude filtering operation involves providing, as the incoming audio signal, samples of the incoming audio signal over a specified window of time. The first and second impulse response functions may be determined from an input signal input into a microphone.

公开(公告)号：US09721582B1

公开(公告)日：2017-08-01

申请号：US15014481

申请日：2016-02-03

Applicant: Google Inc.

Inventor： Yiteng Huang ,  Alejandro Luebs ,  Jan Skoglund ,  Willem Bastiaan Kleijn

IPC: H04R3/00 ,  G10L21/0216 ,  G10L21/0264 ,  G10L21/02 ,  G10L25/21 ,  G10L21/0308

CPC classification number: G10L21/0264 ,  G10L21/0205 ,  G10L21/0216 ,  G10L21/0308 ,  G10L21/0364 ,  G10L25/21 ,  G10L2021/02166 ,  H04R3/005

Abstract: Existing post-filtering methods for microphone array speech enhancement have two common deficiencies. First, they assume that noise is either white or diffuse and cannot deal with point interferers. Second, they estimate the post-filter coefficients using only two microphones at a time, performing averaging over all the microphones pairs, yielding a suboptimal solution. The provided method describes a post-filtering solution that implements signal models which handle white noise, diffuse noise, and point interferers. The method also implements a globally optimized least-squares approach of microphones in a microphone array, providing a more optimal solution than existing conventional methods. Experimental results demonstrate the described method outperforming conventional methods in various acoustic scenarios.

公开(公告)号：US09721580B2

公开(公告)日：2017-08-01

申请号：US14230404

申请日：2014-03-31

Applicant: GOOGLE INC.

Inventor： Jan Skoglund ,  Alejandro Luebs

IPC: G10L21/0208 ,  G10L25/84 ,  G10L25/90 ,  G10L25/78

CPC classification number: G10L21/0208 ,  G10L25/78 ,  G10L25/84 ,  G10L25/90

Abstract: Provided are methods and systems for providing situation-dependent transient noise suppression for audio signals. Different strategies (e.g., levels of aggressiveness) of transient suppression and signal restoration are applied to audio signals associated with participants in a video/audio conference depending on whether or not each participant is speaking (e.g., whether a voiced segment or an unvoiced/non-speech segment of audio is present). If no participants are speaking or there is an unvoiced/non-speech sound present, a more aggressive strategy for transient suppression and signal restoration is utilized. On the other hand, where voiced audio is detected (e.g., a participant is speaking), the methods and systems apply a softer, less aggressive suppression and restoration process.