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    Bi-magnitude processing framework for nonlinear echo cancellation in mobile devices
    1.
    发明授权
    Bi-magnitude processing framework for nonlinear echo cancellation in mobile devices 有权

    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.

    Hierarchical decorrelation of multichannel audio
    3.
    发明授权
    Hierarchical decorrelation of multichannel audio 有权

    公开(公告)号：US10141000B2

    公开(公告)日：2018-11-27

    申请号：US15182751

    申请日：2016-06-15

    Applicant: GOOGLE INC.

    Inventor： Minyue Li ,  Jan Skoglund ,  Willem Bastiaan Kleijn

    IPC: G10L19/24 ,  G10L19/008 ,  G10L19/02 ,  G10L19/035 ,  G10L19/16 ,  G10L19/00

    Abstract: Provided are methods, systems, and apparatus for hierarchical decorrelation of multichannel audio. A hierarchical decorrelation algorithm is designed to adapt to possibly changing characteristics of an input signal, and also preserves the energy of the original signal. The algorithm is invertible in that the original signal can be retrieved if needed. Furthermore, the proposed algorithm decomposes the decorrelation process into multiple low-complexity steps. The contribution of these steps is generally in a decreasing order, and thus the complexity of the algorithm can be scaled.

    Objective speech quality metric
    6.
    发明授权
    Objective speech quality metric 有权
    客观语音质量度量

    公开(公告)号：US09524733B2

    公开(公告)日：2016-12-20

    申请号：US13891978

    申请日：2013-05-10

    Applicant: Google Inc.

    Inventor： Jan Skoglund ,  Andrew J. Hines ,  Noami A. Harte ,  Anil Kokaram

    IPC: G10L25/60

    CPC classification number: G10L25/60

    Abstract: Methods and systems are provided for using a model of human speech quality perception to provide an objective measure for predicting subjective quality assessments. A Virtual Speech Quality Objective Listener (ViSQOL) model is a signal-based full-reference metric that uses a spectro-temporal measure of similarity between a reference signal and test speech signal. Specifically, the model provides for the ability to detect and predict the level of clock drift, and determine whether such clock drift will impact a listener's quality of experience.

    Abstract translation: 提供了使用人类语言质量感知模型的方法和系统来提供用于预测主观质量评估的客观量度。 虚拟语音质量目标监听器（ViSQOL）是一种基于信号的全参考度量，它使用参考信号和测试语音信号之间的相似性的频谱测量。 具体来说，该模型提供了检测和预测时钟漂移水平的能力，并确定这种时钟漂移是否会影响听众的体验质量。

    Keyboard typing detection and suppression
    7.
    发明授权
    Keyboard typing detection and suppression 有权
    键盘打字检测和抑制

    公开(公告)号：US09520141B2

    公开(公告)日：2016-12-13

    申请号：US13781262

    申请日：2013-02-28

    Applicant: GOOGLE INC.

    Inventor： Jens Enzo Nyby Christensen ,  Simon J. Godsill ,  Jan Skoglund

    IPC: G10L21/02 ,  G10L25/48 ,  G10L21/0216 ,  G10L25/84 ,  G10L25/93 ,  G10L21/0208

    CPC classification number: G10L25/48 ,  G10L21/02 ,  G10L21/0208 ,  G10L21/0216 ,  G10L25/84 ,  G10L25/93 ,  G10L2025/935

    Abstract: Provided are methods and systems for detecting the presence of a transient noise event in an audio stream using primarily or exclusively the incoming audio data. Such an approach offers improved temporal resolution and is computationally efficient. The methods and systems presented utilize some time-frequency representation of an audio signal as the basis in a predictive model in an attempt to find outlying transient noise events and interpret the true detection state as a Hidden Markov Model (HMM) to model temporal and frequency cohesion common amongst transient noise events.

    Abstract translation: 提供了用于检测音频流中瞬时噪声事件的存在的方法和系统，其主要或排他地使用输入音频数据。 这种方法提供了改进的时间分辨率，并且在计算上是有效的。 所提出的方法和系统利用音频信号的一些时间频率表示作为预测模型的基础，以试图找出偏离的瞬态噪声事件，并将真实检测状态解释为隐马尔可夫模型（HMM）来模拟时间和频率 瞬态噪声事件中共同的凝聚力。

    Detection of chopped speech
    8.
    发明授权
    Detection of chopped speech 有权
    检测切碎的言语

    公开(公告)号：US09263061B2

    公开(公告)日：2016-02-16

    申请号：US13899381

    申请日：2013-05-21

    Applicant: Google Inc.

    Inventor： Andrew J. Hines ,  Jan Skoglund ,  Naomi Harte ,  Anil Kokaram

    IPC: G10L25/78

    CPC classification number: G10L25/78 ,  G10L21/0232 ,  G10L25/60

    Abstract: Methods and systems are provided for detecting chop in an audio signal. A time-frequency representation, such as a spectrogram, is created for an audio signal and used to calculate a gradient of mean power per frame of the audio signal. Positive and negative gradients are defined for the signal based on the gradient of mean power, and a maximum overlap offset between the positive and negative gradients is determined by calculating a value that maximizes the cross-correlation of the positive and negative gradients. The negative gradient values may be combined (e.g., summed) with the overlap offset, and the combined values then compared with a threshold to estimate the amount of chop present in the audio signal. The chop detection model provided is low-complexity and is applicable to narrowband, wideband, and superwideband speech.

    Abstract translation: 提供了用于检测音频信号中的斩波的方法和系统。 为音频信号创建时频表示，如频谱图，用于计算音频信号每帧平均功率的梯度。 基于平均功率梯度的信号定义正和负梯度，通过计算使正和负梯度的互相关最大化的值来确定正梯度和负梯度之间的最大重叠偏移。 负梯度值可以与重叠偏移组合（例如，相加），然后将组合值与阈值进行比较以估计音频信号中存在的斩波量。 提供的斩波检测模型是低复杂度的，适用于窄带，宽带和超宽带语音。

    POST-PROCESSED REFERENCE PATH FOR ACOUSTIC ECHO CANCELLATION
    9.
    发明申请
    POST-PROCESSED REFERENCE PATH FOR ACOUSTIC ECHO CANCELLATION 审中-公开
    后处理参考路径用于声学ECHO取消

    公开(公告)号：US20150249884A1

    公开(公告)日：2015-09-03

    申请号：US13651893

    申请日：2012-10-15

    Applicant: Google Inc.

    Inventor： Leng Ooi ,  Jan Skoglund ,  Andrew John MacDonald

    IPC: H04R3/00

    CPC classification number: H04R3/02 ,  H04R2499/11

    Abstract: A system includes a speaker, an acoustic echo canceller, a post-processor configured to create a post-processed render signal associated with an audio input, and a reference path operatively connected to the speaker, the post-processor, and the acoustic echo canceller. The reference path provides the acoustic echo canceller with access to the post-processed render signal.

    Abstract translation: 一种系统包括扬声器，声回波消除器，配置成创建与音频输入相关联的后处理渲染信号的后处理器，以及可操作地连接到扬声器，后处理器和声回波消除器的参考路径 。 参考路径提供声学回声消除器，可以访问后处理的渲染信号。

    Incoherent idempotent ambisonics rendering
    10.
    发明授权
    Incoherent idempotent ambisonics rendering 有权

    公开(公告)号：US10015618B1

    公开(公告)日：2018-07-03

    申请号：US15666220

    申请日：2017-08-01

    Applicant: Google Inc.

    Inventor： Willem Bastiaan Kleijn ,  Andrew Allen ,  Jan Skoglund ,  Sze Chie Lim

    IPC: H04R5/02 ,  H04S7/00 ,  H04S3/00

    CPC classification number: H04S7/303 ,  H04S3/008 ,  H04S3/02 ,  H04S2400/01 ,  H04S2400/11 ,  H04S2420/11

    Abstract: Techniques of rendering sound for a listener involve producing, as the amplitude of each of the source driving signals, a sum of two terms: a first term based on a solution s† to the equation b=A·s, and a second term based on a projection of a specified vector ŝ onto the nullspace of A, ŝ not being a solution to the equation b=A·s. Along these lines, in one example, the first term is equivalent to a Moore-Penrose pseudoinverse, e.g., AH(AAH)−1·b. In general, any solution to the equation b=A·s is satisfactory. The specified vector that is projected onto the nullspace of A is defined to reduce the coherence of the net sound field. Advantageously, the resulting operator is both linear time-invariant and idempotent so that the sound field may be faithfully reproduce both inside the RSF and at a sufficient range outside the RSF to cover a human head.

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