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    VOICE QUALITY ENHANCEMENT TECHNIQUES, SPEECH RECOGNITION TECHNIQUES, AND RELATED SYSTEMS
    1.
    发明申请
    VOICE QUALITY ENHANCEMENT TECHNIQUES, SPEECH RECOGNITION TECHNIQUES, AND RELATED SYSTEMS 有权
    语音质量增强技术，语音识别技术及相关系统

    公开(公告)号：US20150112672A1

    公开(公告)日：2015-04-23

    申请号：US14517700

    申请日：2014-10-17

    Applicant: Apple Inc.

    Inventor： Daniele Giacobello ,  Jason Wung ,  Joshua Atkins ,  Ramin Pichevar ,  Raghavendra Prabhu

    IPC: G10L21/0208 ,  G10L15/20

    CPC classification number: G10L21/0208 ,  H04M9/082

    Abstract: An echo canceller can be arranged to receive an input signal and to receive a reference signal. The echo canceller can subtract a linear component of the reference signal from the input signal. A noise suppressor can suppress non-linear effects of the reference signal in the input signal in correspondence with a large number of selectable parameters. Such suppression can be provided on a frequency-by-frequency basis, with a unique set of tunable parameters selected for each frequency. A degree of suppression provided by the noise suppressor can correspond to an estimate of residual echo remaining after the one or more linear components of the reference signal have been subtracted from the input signal, to an estimated double-talk probability, and to an estimated signal-to-noise ratio of near-end speech in the input signal for each respective frequency. A speech recognizer can receive a processed input signal from the noise suppressor.

    Abstract translation: 可以设置回波消除器以接收输入信号并接收参考信号。 回波消除器可以从输入信号中减去参考信号的线性分量。 噪声抑制器可以抑制与大量可选参数对应的输入信号中的参考信号的非线性效应。 可以在频率基础上提供这种抑制，并为每个频率选择一组唯一的可调参数。 由噪声抑制器提供的抑制程度可以对应于从输入信号中减去参考信号的一个或多个线性分量之后剩余回波的估计值，估计的双方通话概率，以及估计的信号 每个相应频率的输入信号中的近端语音的噪声比。 语音识别器可以从噪声抑制器接收经处理的输入信号。

    System and method for performing speech enhancement using a deep neural network-based signal
    3.
    发明授权
    System and method for performing speech enhancement using a deep neural network-based signal 有权

    公开(公告)号：US10074380B2

    公开(公告)日：2018-09-11

    申请号：US15227885

    申请日：2016-08-03

    Applicant: Apple Inc.

    Inventor： Jason Wung ,  Ramin Pishehvar ,  Daniele Giacobello ,  Joshua D. Atkins

    IPC: G10L21/02 ,  G10L21/0232 ,  G10L25/30 ,  G10L25/87 ,  G10L21/0208

    CPC classification number: G10L21/0232 ,  G10L25/30 ,  G10L25/87 ,  G10L2021/02082

    Abstract: Method for performing speech enhancement using a Deep Neural Network (DNN)-based signal starts with training DNN offline by exciting a microphone using target training signal that includes signal approximation of clean speech. Loudspeaker is driven with a reference signal and outputs loudspeaker signal. Microphone then generates microphone signal based on at least one of: near-end speaker signal, ambient noise signal, or loudspeaker signal. Acoustic-echo-canceller (AEC) generates AEC echo-cancelled signal based on reference signal and microphone signal. Loudspeaker signal estimator generates estimated loudspeaker signal based on microphone signal and AEC echo-cancelled signal. DNN receives microphone signal, reference signal, AEC echo-cancelled signal, and estimated loudspeaker signal and generates a speech reference signal that includes signal statistics for residual echo or for noise. Noise suppressor generates a clean speech signal by suppressing noise or residual echo in the microphone signal based on speech reference signal. Other embodiments are described.

    End-To-End Time-Domain Multitask Learning for ML-Based Speech Enhancement
    4.
    发明申请
    End-To-End Time-Domain Multitask Learning for ML-Based Speech Enhancement 有权

    公开(公告)号：US20220366927A1

    公开(公告)日：2022-11-17

    申请号：US17321411

    申请日：2021-05-15

    Applicant: Apple Inc.

    Inventor： Ramin Pishehvar ,  Ante Jukic ,  Mehrez Souden ,  Jason Wung ,  Feipeng Li ,  Joshua D. Atkins

    IPC: G10L21/0216 ,  G10L15/16 ,  G06N20/00

    Abstract: Disclosed is a multi-task machine learning model such as a time-domain deep neural network (DNN) that jointly generate an enhanced target speech signal and target audio parameters from a mixed signal of target speech and interference signal. The DNN may encode the mixed signal, determine masks used to jointly estimate the target signal and the target audio parameters based on the encoded mixed signal, apply the mask to separate the target speech from the interference signal to jointly estimate the target signal and the target audio parameters, and decode the masked features to enhance the target speech signal and to estimate the target audio parameters. The target audio parameters may include a voice activity detection (VAD) flag of the target speech. The DNN may leverage multi-channel audio signal and multi-modal signals such as video signals of the target speaker to improve the robustness of the enhanced target speech signal.

    DEEP LEARNING DRIVEN MULTI-CHANNEL FILTERING FOR SPEECH ENHANCEMENT
    5.
    发明申请
    DEEP LEARNING DRIVEN MULTI-CHANNEL FILTERING FOR SPEECH ENHANCEMENT 审中-公开

    公开(公告)号：US20190172476A1

    公开(公告)日：2019-06-06

    申请号：US15830955

    申请日：2017-12-04

    Applicant: Apple Inc.

    Inventor： Jason Wung ,  Mehrez Souden ,  Ramin Pishehvar ,  Joshua D. Atkins

    IPC: G10L21/02 ,  G10L25/30 ,  G10L15/02 ,  G10L21/0232 ,  G10L25/03

    Abstract: A number of features are extracted from a current frame of a multi-channel speech pickup and from side information that is a linear echo estimate, a diffuse signal component, or a noise estimate of the multi-channel speech pickup. A DNN-based speech presence probability is produced for the current frame, where the SPP value is produced in response to the extracted features being input to the DNN. The DNN-based SPP value is applied to configure a multi-channel filter whose input is the multi-channel speech pickup and whose output is a single audio signal. In one aspect, the system is designed to run online, at low enough latency for real time applications such voice trigger detection. Other aspects are also described and claimed.

    Voice quality enhancement techniques, speech recognition techniques, and related systems
    6.
    发明授权
    Voice quality enhancement techniques, speech recognition techniques, and related systems 有权

    公开(公告)号：US09633671B2

    公开(公告)日：2017-04-25

    申请号：US14517700

    申请日：2014-10-17

    Applicant: Apple Inc.

    Inventor： Daniele Giacobello ,  Jason Wung ,  Joshua Atkins ,  Ramin Pichevar ,  Raghavendra Prabhu

    IPC: G10L15/20 ,  G10L21/0208 ,  H04M9/08

    CPC classification number: G10L21/0208 ,  H04M9/082

    Abstract: An echo canceller can be arranged to receive an input signal and to receive a reference signal. The echo canceller can subtract a linear component of the reference signal from the input signal. A noise suppressor can suppress non-linear effects of the reference signal in the input signal in correspondence with a large number of selectable parameters. Such suppression can be provided on a frequency-by-frequency basis, with a unique set of tunable parameters selected for each frequency. A degree of suppression provided by the noise suppressor can correspond to an estimate of residual echo remaining after the one or more linear components of the reference signal have been subtracted from the input signal, to an estimated double-talk probability, and to an estimated signal-to-noise ratio of near-end speech in the input signal for each respective frequency. A speech recognizer can receive a processed input signal from the noise suppressor.

    Echo cancellation using a subset of multiple microphones as reference channels
    8.
    发明授权
    Echo cancellation using a subset of multiple microphones as reference channels 有权

    公开(公告)号：US10978086B2

    公开(公告)日：2021-04-13

    申请号：US16517400

    申请日：2019-07-19

    Applicant: Apple Inc.

    Inventor： Jason Wung ,  Sarmad Aziz Malik ,  Ashrith Deshpande ,  Ante Jukic ,  Joshua D. Atkins

    IPC: G10L21/0208 ,  G10K11/178 ,  G10L21/0216

    Abstract: An echo canceller is disclosed in which audio signals of the playback content received by one or more of the microphones from a loudspeaker of the device may be used as the playback reference signals to estimate the echo signals of the playback content received by a target microphone for echo cancellation. The echo canceller may estimate the transfer function between a reference microphone and the target microphone based on the playback reference signal of the reference microphone and the signal of the target microphone. To mitigate near-end speech cancellation at the target microphone, the echo canceller may compute a mask to distinguish between target microphone audio signals that are echo-signal dominant and near-end speech dominant. The echo canceller may use the mask to adaptively update the transfer function or to modify the playback reference signal used by the transfer function to estimate the echo signals of the playback content.

    Echo Cancellation Using A Subset of Multiple Microphones As Reference Channels
    9.
    发明申请
    Echo Cancellation Using A Subset of Multiple Microphones As Reference Channels 有权

    公开(公告)号：US20210020188A1

    公开(公告)日：2021-01-21

    申请号：US16517400

    申请日：2019-07-19

    Applicant: Apple Inc.

    Inventor： Jason Wung ,  Sarmad Aziz Malik ,  Ashrith Deshpande ,  Ante Jukic ,  Joshua D. Atkins

    IPC: G10L21/0208 ,  G10K11/178

    Abstract: An echo canceller is disclosed in which audio signals of the playback content received by one or more of the microphones from a loudspeaker of the device may be used as the playback reference signals to estimate the echo signals of the playback content received by a target microphone for echo cancellation. The echo canceller may estimate the transfer function between a reference microphone and the target microphone based on the playback reference signal of the reference microphone and the signal of the target microphone. To mitigate near-end speech cancellation at the target microphone, the echo canceller may compute a mask to distinguish between target microphone audio signals that are echo-signal dominant and near-end speech dominant. The echo canceller may use the mask to adaptively update the transfer function or to modify the playback reference signal used by the transfer function to estimate the echo signals of the playback content.

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