公开(公告)号：US20170337932A1

公开(公告)日：2017-11-23

申请号：US15159698

申请日：2016-05-19

Applicant: Apple Inc.

Inventor： Vasu Iyengar ,  Ashrith Deshpande ,  Aram M. Lindahl

IPC: G10L21/0208 ,  G10L25/78

Abstract: An audio system has a housing in which are integrated a number of microphones. A programmed processor accesses the microphone signals and produces a number of acoustic pick up beams. A number of separation values are computed, each being a measure of the difference between strength of a respective beam and strength of a noise reference input signal. One of the beams is selected whose separation value is the largest, and the selected beam is applied to a first input of a two-channel noise suppression process, while the noise reference input signal is applied to the second input of the noise suppression process. Other embodiments are also described and claimed.

公开(公告)号：US09491545B2

公开(公告)日：2016-11-08

申请号：US14286659

申请日：2014-05-23

Applicant: Apple Inc.

Inventor： Vasu Iyengar ,  Martin E. Johnson ,  Ronald N. Isaac ,  Aram M. Lindahl

IPC: H04R3/00 ,  H04M9/08 ,  H04R3/04 ,  H04S7/00 ,  G10L21/0208 ,  H04B3/23

CPC classification number: H04R3/005 ,  G10L2021/02082 ,  H04B3/23 ,  H04M9/082 ,  H04R3/04 ,  H04R2410/05 ,  H04R2499/11 ,  H04S7/305

Abstract: In one embodiment, a process for suppressing reverberation begins with a device of a user obtaining a reverberant speech signal from a voice of the user. The device determines a first estimated reverberation component of the reverberant speech signal. The device generates a first de-reverberated output signal with a first reverberation suppression based on the reverberant speech signal and the first estimated reverberation component. Then, the device generates a second improved reverberation component using the first de-reverberated output signal. The device generates a second de-reverberated output signal with a second reverberation suppression based on the reverberant speech signal and the second improved reverberation component.

Abstract translation: 在一个实施例中，用于抑制混响的过程开始于用户的设备从用户的语音获得混响语音信号。 该装置确定混响语音信号的第一估计混响分量。 该装置基于混响语音信号和第一估计混响分量产生具有第一混响抑制的第一去混响输出信号。 然后，该装置使用第一去混响输出信号产生第二改进的混响分量。 该装置基于混响语音信号和第二改进的混响分量产生具有第二混响抑制的第二去混响输出信号。

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。

公开(公告)号：US11276384B2

公开(公告)日：2022-03-15

申请号：US16871661

申请日：2020-05-11

Applicant: Apple Inc.

Inventor： John Woodruff ,  Tom-Davy W. Saux ,  Vladan Bajic ,  Vasu Iyengar ,  Andrew E. Greenwood ,  Tyrone T. Chen ,  Jakub Mazur ,  Tony S. Verma

IPC: G10K11/178

Abstract: An audio system has an ambient sound enhancement function, in which an against-the-ear audio device having a speaker converts a digitally processed version of an input audio signal into sound. The audio system also has an acoustic noise cancellation (ANC) function that may be combined in various ways with the sound enhancement function, and that may be responsive to voice activity detection. Other aspects are also described and claimed.

公开(公告)号：US10396835B2

公开(公告)日：2019-08-27

申请号：US15625999

申请日：2017-06-16

Applicant: Apple Inc.

Inventor： Ruchir M. Dave ,  Ashrith Deshpande ,  Vasu Iyengar

IPC: H04M1/02 ,  H04B1/04 ,  H04M1/60 ,  H04J3/14

Abstract: Systems and methods for reducing effects of time-division multiplexing noise in mobile communications devices. When cellular communication with time-division multiplexing is detected, such as Global System for Mobiles (GSM) communication with Time Division Multiple Access (TDMA) protocol, total energy and energy at a repetition frequency of the time division multiplexing is measured in audio signals received from several microphones located in the device. A control signal indicating microphones affected by TDMA noise is provided to signal processing subsystems that receive audio signals from the microphones. A beam former circuit may combine two or more audio signals to produce beam formed signals. The control signal may further indicate beam formed signals affected by TDMA noise based on a ratio of the energy from the repetition frequency to the total energy in the beam formed signals.

公开(公告)号：US10382857B1

公开(公告)日：2019-08-13

申请号：US15939224

申请日：2018-03-28

Applicant: Apple Inc.

Inventor： Sarthak Khanal ,  Hannes Breitschaedel ,  Vasu Iyengar

IPC: H04R1/22 ,  H04R3/04

Abstract: An audio electronics system operates on audio data. A low-pass or bandpass filter produces first data from audio data. A level detector produces a time-varying first gain. The first gain is based on a time-varying level of the first data. A harmonics generator receives, as input, the first data adjusted by an inverse of the first gain. The harmonics generator produces second data, as harmonics of the input. A multiplier outputs the second data adjusted by the first gain. Other aspects are also described and claimed.

公开(公告)号：US20190066710A1

公开(公告)日：2019-02-28

申请号：US15688455

申请日：2017-08-28

Applicant: Apple Inc.

Inventor： Nicholas J. Bryan ,  Vasu Iyengar ,  Aram M. Lindahl

IPC: G10L21/02 ,  G10L15/22 ,  G10L21/0272 ,  G10L21/0232 ,  G10L15/30 ,  G10L25/51 ,  G10L21/0316

Abstract: A method for controlling a speech enhancement process in a far-end device, while engaged in a voice or video telephony communication session over a communication link with a near-end device. A near-end user speech signal is produced, using a microphone to pick up speech of a near-end user, and is analyzed by an automatic speech recognizer (ASR) without being triggered by an ASR trigger phrase or button. The recognized words are compared to a library of phrases to select a matching phrase, where each phrase is associated with a message that represents an audio signal processing operation. The message associated with the matching phrase is sent to the far-end device, which is used to configure the far-end device to adjust the speech enhancement process that produces the far-end speech signal. Other embodiments are also described.

公开(公告)号：US10176823B2

公开(公告)日：2019-01-08

申请号：US14274544

申请日：2014-05-09

Applicant: Apple Inc.

Inventor： Sorin V. Dusan ,  Aram M. Lindahl ,  Alexander Kanaris ,  Vasu Iyengar

IPC: G10L21/0216 ,  G10L21/0208

Abstract: Electronic system for audio noise processing and noise reduction comprises: first and second noise estimators, selector and attenuator. First noise estimator processes first audio signal from voice beamformer (VB) and generate first noise estimate. VB generates first audio signal by beamforming audio signals from first and second audio pick-up channels. Second noise estimator processes first and second audio signal from noise beamformer (NB), in parallel with first noise estimator and generates second noise estimate. NB generates second audio signal by beamforming audio signals from first and second audio pick-up channels. First and second audio signals include frequencies in first and second frequency regions. Selector's output noise estimate may be a) second noise estimate in the first frequency region, and b) first noise estimate in the second frequency region. Attenuator attenuates first audio signal in accordance with output noise estimate. Other embodiments are also described.

公开(公告)号：US20180367172A1

公开(公告)日：2018-12-20

申请号：US15625999

申请日：2017-06-16

Applicant: Apple Inc.

Inventor： Ruchir M. Dave ,  Ashrith Deshpande ,  Vasu Iyengar

IPC: H04B1/04

CPC classification number: H04B1/0475 ,  H04J3/14 ,  H04M1/0202 ,  H04M1/6008

Abstract: Systems and methods for reducing effects of time-division multiplexing noise in mobile communications devices. When cellular communication with time-division multiplexing is detected, such as Global System for Mobiles (GSM) communication with Time Division Multiple Access (TDMA) protocol, total energy and energy at a repetition frequency of the time division multiplexing is measured in audio signals received from several microphones located in the device. A control signal indicating microphones affected by TDMA noise is provided to signal processing subsystems that receive audio signals from the microphones. A beam former circuit may combine two or more audio signals to produce beam formed signals. The control signal may further indicate beam formed signals affected by TDMA noise based on a ratio of the energy from the repetition frequency to the total energy in the beam formed signals.

公开(公告)号：US20160360314A1

公开(公告)日：2016-12-08

申请号：US14732770

申请日：2015-06-07

Applicant: Apple Inc.

Inventor： Vasu Iyengar ,  Joshua D. Atkins ,  Aram M. Lindahl ,  Tarun Pruthi ,  Ashrith Deshpande

IPC: H04R1/40 ,  G10L21/02 ,  G10L21/0216 ,  H04M1/03

CPC classification number: H04R1/406 ,  G10L21/0208 ,  G10L2021/02166 ,  H04R3/005 ,  H04R2430/20 ,  H04R2499/11

Abstract: An orientation detector can have a first microphone, a second microphone, and a reference microphone spaced from the first microphone and the second microphone. An orientation processor can be configured to determine an orientation of the first microphone, the second microphone, or both, relative to a user's mouth based on a comparison of a relative strength of a first signal associated with the first microphone to a relative strength of a second signal associated with the second microphone. A channel selector in a speech enhancer can select one signal from among several signals based at least in part on the orientation determined by the orientation processor. A mobile communication handset can include a microphone-based orientation detector of the type disclosed herein.

Abstract translation: 取向检测器可以具有与第一麦克风和第二麦克风间隔开的第一麦克风，第二麦克风和参考麦克风。 取向处理器可以被配置为基于与第一麦克风相关联的第一信号的相对强度与第一麦克风的相对强度的比较来相对于用户的嘴来确定第一麦克风，第二麦克风或两者的取向 第二信号与第二麦克风相关联。 语音增强器中的频道选择器可以至少部分地基于由取向处理器确定的取向，从多个信号中选择一个信号。 移动通信手机可以包括本文公开的类型的基于麦克风的定向检测器。