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    AUDIO ADAPTATION TO ROOM
    31.
    发明申请
    AUDIO ADAPTATION TO ROOM 审中-公开

    公开(公告)号：US20180352333A1

    公开(公告)日：2018-12-06

    申请号：US15636967

    申请日：2017-06-29

    Applicant: Apple Inc.

    Inventor： Adam E. Kriegel ,  Afrooz Family ,  Sean A. Ramprashad ,  Sylvain J. Choisel

    IPC: H04R3/04 ,  G10L21/0208 ,  H04R1/40 ,  H04R29/00 ,  H04R3/12

    CPC classification number: H04R3/04 ,  G10L21/0208 ,  G10L2021/02082 ,  H04R1/403 ,  H04R3/12 ,  H04R27/00 ,  H04R29/002 ,  H04R29/007 ,  H04R2227/005 ,  H04S7/305 ,  H04S7/307

    Abstract: An audio system includes one or more loudspeaker cabinets, each having loudspeakers. The system outputs an omnidirectional sound pattern to determine the acoustic environment. Sensing logic determines an acoustic environment of the loudspeaker cabinets. The sensing logic may include an echo canceller. A playback mode processor adjusts an audio program according to a playback mode determined from the acoustic environment of the audio system. The system may produce a directional pattern superimposed on an omnidirectional pattern, if the acoustic environment is in free space. The system may aim ambient content toward a wall and direct content away from the wall, if the acoustic environment is not in free space. The sensing logic automatically determines the acoustic environment upon initial power up and when position changes of loudspeaker cabinets are detected. Accelerometers may detect position changes of the loudspeaker cabinets.

    Systems and methods for determining the condition of multiple microphones
    32.
    发明授权
    Systems and methods for determining the condition of multiple microphones 有权
    用于确定多个麦克风状况的系统和方法

    公开(公告)号：US09432787B2

    公开(公告)日：2016-08-30

    申请号：US15019521

    申请日：2016-02-09

    Applicant: Apple Inc.

    Inventor： Arvindh Krishnaswamy ,  David T. Yeh ,  Juha O. Merimaa ,  Sean A. Ramprashad

    IPC: H04R29/00 ,  H04R3/00

    CPC classification number: H04R29/005 ,  H04R3/00 ,  H04R29/004 ,  H04R2499/11

    Abstract: Systems and methods for determining the operating condition of multiple microphones of an electronic device are disclosed. A system can include a plurality of microphones operative to receive signals, a microphone condition detector, and a plurality of microphone condition determination sources. The microphone condition detector can determine a condition for each of the plurality of microphones by using the received signals and accessing at least one microphone condition determination source.

    Abstract translation: 公开了用于确定电子设备的多个麦克风的操作条件的系统和方法。 系统可以包括可操作以接收信号的多个麦克风，麦克风条件检测器和多个麦克风条件确定源。 麦克风条件检测器可以通过使用接收到的信号并访问至少一个麦克风条件确定源来确定多个麦克风中的每一个的条件。

    Psychoacoustic adaptive notch filtering
    33.
    发明授权
    Psychoacoustic adaptive notch filtering 有权
    心理声学自适应陷波滤波

    公开(公告)号：US09386371B2

    公开(公告)日：2016-07-05

    申请号：US13790666

    申请日：2013-03-08

    Applicant: Apple Inc.

    Inventor： Arvindh Krishnaswamy ,  Sean A. Ramprashad

    IPC: H04B15/00 ,  H04R3/00 ,  G10L21/0324 ,  H04R5/04

    CPC classification number: H04R3/002 ,  G10L21/0324 ,  H04R3/00 ,  H04R5/04

    Abstract: Improved systems and methods for psychoacoustic adaptive notch filtering are provided. By accounting for psychoacoustic properties of an audio signal as well as finer characteristics of noise which may be present in the audio signal (e.g., the shape of the spectral density of the noise), more effective strategies for dealing with undesirable components of the audio signal may be realized.

    Abstract translation: 提供了用于心理声学自适应陷波滤波的改进的系统和方法。 通过考虑音频信号的心理声学特性以及可能存在于音频信号中的更精细的噪声特性（例如，噪声的频谱密度的形状），用于处理音频信号的不期望的分量的更有效的策略 可以实现。

    SYSTEM AND METHOD OF DOUBLE TALK DETECTION WITH ACOUSTIC ECHO AND NOISE CONTROL
    34.
    发明申请
    SYSTEM AND METHOD OF DOUBLE TALK DETECTION WITH ACOUSTIC ECHO AND NOISE CONTROL 有权
    双声道检测系统和方法与声学和噪声控制

    公开(公告)号：US20160127535A1

    公开(公告)日：2016-05-05

    申请号：US14880824

    申请日：2015-10-12

    Applicant: Apple Inc.

    Inventor： Lalin S. Theverapperuma ,  Vasu Iyengar ,  Sean A. Ramprashad

    IPC: H04M3/00 ,  H04L5/14 ,  G10L21/0264 ,  H04M1/60 ,  G10L21/0208 ,  G10L21/02

    CPC classification number: H04M3/002 ,  G10L21/0205 ,  G10L21/0208 ,  G10L21/0264 ,  G10L2021/02082 ,  G10L2021/02161 ,  G10L2021/02165 ,  H04L5/14 ,  H04M1/60 ,  H04M1/6041 ,  H04M9/082

    Abstract: System of improving sound quality includes loudspeaker, microphone, accelerometer, acoustic-echo-cancellers (AEC), and double-talk detector (DTD). Loudspeaker outputs loudspeaker signal including downlink audio signal from far-end speaker. Microphone generates microphone uplink signal and receives at least one of: near-end speaker, ambient noise, and loudspeaker signals. Accelerometer generates accelerometer-uplink signal and receives at least one of: near-end speaker, ambient noise, and loudspeaker signals. First AEC receives downlink audio, microphone-uplink and double talk control signals, and generates AEC-microphone linear echo estimate and corrected AEC-microphone uplink signal. Second AEC receives downlink audio, accelerometer uplink and double talk control signals, and generates AEC-accelerometer linear echo estimate and corrected AEC-accelerometer uplink signal. DTD receives downlink audio signal, uplink signals, corrected uplink signals, linear echo estimates, and generates double-talk control signal. Uplink audio signal including at least one of corrected microphone-uplink signal and corrected accelerometer-uplink signal is generated. Other embodiments are described.

    Abstract translation: 提高音质的系统包括扬声器，麦克风，加速度计，声回波消除器（AEC）和双通话检测器（DTD）。 扬声器输出包括来自远端扬声器的下行音频信号的扬声器信号。 麦克风产生麦克风上行链路信号，并接收以下至少一个：近端扬声器，环境噪声和扬声器信号。 加速度计产生加速度计 - 上行链路信号并接收以下至少一种：近端扬声器，环境噪声和扬声器信号。 第一AEC接收下行音频，麦克风上行和双通话控制信号，并生成AEC麦克风线性回波估计和校正的AEC麦克风上行信号。 第二个AEC接收下行音频，加速度计上行链路和双通道控制信号，并生成AEC加速度计线性回波估计和校正的AEC加速度计上行信号。 DTD接收下行链路音频信号，上行链路信号，校正的上行链路信号，线性回波估计，并产生双向对话控制信号。 产生包括校正的麦克风 - 上行链路信号和校正的加速度计 - 上行链路信号中的至少一个的上行链路音频信号。 描述其他实施例。

    SPATIAL AUDIO CONTROLLER
    38.
    发明申请
    SPATIAL AUDIO CONTROLLER 有权

    公开(公告)号：US20220394407A1

    公开(公告)日：2022-12-08

    申请号：US17339864

    申请日：2021-06-04

    Applicant: Apple Inc.

    Inventor： Sean A. Ramprashad ,  Peter D. Callaway ,  Jae Woo Chang ,  Martin E. Johnson ,  Daniel K. Boothe ,  Kostyantyn Komarov ,  Patrick Miauton ,  Christopher M. Garrido ,  Austin W. Shyu ,  Karthick Santhanam

    IPC: H04S3/00 ,  H04S7/00 ,  G06F3/0487 ,  H04S5/00 ,  H04R3/00

    Abstract: A method performed a local device that is communicatively coupled with several remote devices, the method includes: receiving, from each remote device with which the local device is engaged in a communication session, an input audio stream; receiving, for each remote device, a set parameters; determining, for each input audio stream, whether the input audio stream is to be 1) rendered individually or 2) rendered as a mix of input audio streams based on the set of parameters; for each input audio stream that is determined to be rendered individually, spatial rendering the input audio stream as an individual virtual sound source that contains only that input audio stream; and for input audio streams that are determined to be rendered as the mix of input audio streams, spatial rendering the mix of input audio streams as a single virtual sound source that contains the mix of input audio streams.

    Speech model-based neural network-assisted signal enhancement
    40.
    发明授权
    Speech model-based neural network-assisted signal enhancement 有权

    公开(公告)号：US10381020B2

    公开(公告)日：2019-08-13

    申请号：US15625966

    申请日：2017-06-16

    Applicant: Apple Inc.

    Inventor： Sean A. Ramprashad

    IPC: G10L15/20 ,  G10L19/00 ,  G10L25/15 ,  G10L25/30 ,  G10L21/003 ,  G10L21/0208 ,  G10L21/0232

    Abstract: Several embodiments of a digital speech signal enhancer are described that use an artificial neural network that produces clean speech coding parameters based on noisy speech coding parameters as its input features. A vocoder parameter generator produces the noisy speech coding parameters from a noisy speech signal. A vocoder model generator processes the clean speech coding parameters into estimated clean speech spectral magnitudes. In one embodiment, a magnitude modifier modifies an original frequency spectrum of the noisy speech signal using the estimated clean speech spectral magnitudes, to produce an enhanced frequency spectrum, and a synthesis block converts the enhanced frequency spectrum into time domain, as an output speech sequence. Other embodiments are also described.

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