公开(公告)号：US09510094B2

公开(公告)日：2016-11-29

申请号：US14248834

申请日：2014-04-09

Applicant: Apple Inc.

Inventor： Baptiste P. Paquier ,  Bryan J. James ,  Aram M. Lindahl

IPC: H04R3/00 ,  G10L21/0216

CPC classification number: H04R3/005 ,  G10L21/0205 ,  G10L21/0216 ,  G10L2021/02165 ,  H04R2410/01 ,  H04R2410/05 ,  H04R2430/03 ,  H04R2499/11

Abstract: A mobile device uses externals microphone signals to improve the estimate of background noise that it computes. In order to improve voice quality in a first signal that is produced by an internal microphone, the mobile device identifies an external microphone device within proximity of the mobile device. The mobile device establishes a wireless connection with the external microphone device. The mobile device receives a second signal from the external microphone device through the wireless connection. The second signal is produced by a microphone of the external microphone device. The mobile device generates a noise profile based on the second signal, and then suppresses background/ambient noise from the first signal based on the noise profile. Other embodiments are also described.

Abstract translation: 移动设备使用外部麦克风信号来改善其计算的背景噪声的估计。 为了提高由内部麦克风产生的第一信号中的语音质量，移动设备识别移动设备附近的外部麦克风设备。 移动设备与外部麦克风设备建立无线连接。 移动设备通过无线连接从外部麦克风设备接收第二信号。 第二信号由外部麦克风装置的麦克风产生。 移动设备基于第二信号生成噪声分布，然后基于噪声分布来抑制来自第一信号的背景/环境噪声。 还描述了其它实施例。

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：US11900936B2

公开(公告)日：2024-02-13

申请号：US17733828

申请日：2022-04-29

Applicant: Apple Inc.

Inventor： Aram M. Lindahl

IPC: G10L15/22 ,  G06F16/43 ,  G10L15/30 ,  G06F3/16 ,  G10L21/06 ,  G10L15/18

CPC classification number: G10L15/22 ,  G06F3/167 ,  G06F16/43 ,  G10L15/1822 ,  G10L15/30 ,  G10L21/06 ,  G10L2015/223 ,  G10L2015/228

Abstract: An electronic device may capture a voice command from a user. The electronic device may store contextual information about the state of the electronic device when the voice command is received. The electronic device may transmit the voice command and the contextual information to computing equipment such as a desktop computer or a remote server. The computing equipment may perform a speech recognition operation on the voice command and may process the contextual information. The computing equipment may respond to the voice command. The computing equipment may also transmit information to the electronic device that allows the electronic device to respond to the voice command.

公开(公告)号：US11736862B1

公开(公告)日：2023-08-22

申请号：US17407069

申请日：2021-08-19

Applicant: Apple Inc.

Inventor： Aram M. Lindahl

IPC: H04R3/12 ,  H04R5/04 ,  H04R5/033 ,  H04R5/02 ,  G06F3/04817 ,  G06T13/00 ,  H04S1/00 ,  G06F3/0482 ,  H04S7/00

CPC classification number: H04R3/12 ,  G06F3/0482 ,  G06F3/04817 ,  G06T13/00 ,  H04R5/02 ,  H04R5/033 ,  H04R5/04 ,  H04S1/002 ,  H04S7/304 ,  H04S2420/01

Abstract: An audio system and method of using the audio system to augment spatial audio rendition is described. The audio system can include a device to receive user inputs designating positions on an augmented reality view of a listening environment. Sound source icons can be presented in the augmented reality view at the designated positions. The sound source icons can visually represent sound sources at locations in the listening environment that correspond to, but are different than, the positions. One or more processors of the audio system can apply head-related transfer functions, which correspond to the locations in the listening environment, to audio input signals to generate binaural audio signals. The audio system can include a headset that uses the binaural audio signals to render spatialized audio localizing sounds to the locations in the listening environment. Other aspects are also described and claimed.

公开(公告)号：US20200245088A1

公开(公告)日：2020-07-30

申请号：US16778634

申请日：2020-01-31

Applicant: Apple Inc.

Inventor： Bruce C. Po ,  Richard M. Powell ,  Aram M. Lindahl ,  Baptiste P. Paquier ,  Phillip A. Kamp

IPC: H04S7/00 ,  H04R1/40 ,  H04R3/12 ,  G06F3/16 ,  H04R3/04 ,  H04R5/04 ,  H04R29/00

Abstract: A system and method is described for determining whether a loudspeaker device has relocated, tilted, rotated, or changed environment such that one or more parameters for driving the loudspeaker may be modified and/or a complete reconfiguration of the loudspeaker system may be performed. In one embodiment, the system may include a set of sensors. The sensors provide readings that are analyzed to determine 1) whether the loudspeaker has moved since a previous analysis and/or 2) a distance of movement and/or a degree change in orientation of the loudspeaker since the previous analysis. Upon determining the level of movement is below a threshold value, the system adjusts previous parameters used to drive one or more of the loudspeakers. By adjusting previous parameters instead of performing a complete recalibration, the system provides a more efficient technique for ensuring that the loudspeakers continue to produce accurate sound for the listener.

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：US09886966B2

公开(公告)日：2018-02-06

申请号：US14536269

申请日：2014-11-07

Applicant: Apple Inc.

Inventor： Sorin V. Dusan ,  Aram M. Lindahl

IPC: G10L15/00 ,  B25J5/00 ,  H04N7/14 ,  G10L21/00 ,  H04M1/00 ,  G10L21/0208 ,  G10L15/20 ,  H04N7/15 ,  B25J13/00

CPC classification number: G10L21/0208 ,  B25J13/00 ,  G10L15/20 ,  H04N7/15

Abstract: Method for improving noise suppression for ASR starts with a microphone receiving an audio signal including speech signal and noise signal. In each frame for frequency band of audio signal, a noise estimator detects ambient noise level and generates noise estimate value based on estimated ambient noise level, variable noise suppression target controller generates suppression target value using noise estimate value and logistic function, a gain value calculator generates a gain value based on suppression target value and noise estimate value, and combiner enhances the audio signal by the gain value to generate a clean audio signal in each frame for all frequency bands. Logistic function models desired noise suppression level that varies based on ambient noise level. Variable level of noise suppression includes low attenuation for low noise levels and progressively higher attenuation for higher noise level. Other embodiments are also described.

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。

公开(公告)号：US20160358619A1

公开(公告)日：2016-12-08

申请号：US14732715

申请日：2015-06-06

Applicant: Apple Inc.

Inventor： Sean A. Ramprashad ,  Harvey D. Thornburg ,  Arvindh Krishnaswamy ,  Aram M. Lindahl

IPC: G10L21/0308 ,  G10L15/34 ,  G10L15/16

CPC classification number: G10L15/34 ,  G10L15/16 ,  G10L15/20 ,  G10L2015/022 ,  G10L2021/02166

Abstract: A speech recognition system for resolving impaired utterances can have a speech recognition engine configured to receive a plurality of representations of an utterance and concurrently to determine a plurality of highest-likelihood transcription candidates corresponding to each respective representation of the utterance. The recognition system can also have a selector configured to determine a most-likely accurate transcription from among the transcription candidates. As but one example, the plurality of representations of the utterance can be acquired by a microphone array, and beamforming techniques can generate independent streams of the utterance across various look directions using output from the microphone array.

Abstract translation: 用于解决受损话语的语音识别系统可以具有语音识别引擎，其被配置为接收话语的多个表示，并且同时确定与发音的每个相应表示相对应的多个最高似然转录候选。 识别系统还可以具有选择器，其被配置为确定转录候选物中最可能的精确转录。 作为一个示例，可以通过麦克风阵列来获取话音的多个表示，并且波束成形技术可以使用来自麦克风阵列的输出在各种外观方向上产生独立的话音流。