SYSTEM AND METHOD FOR AUDIO NOISE PROCESSING AND NOISE REDUCTION
    51.
    发明申请
    SYSTEM AND METHOD FOR AUDIO NOISE PROCESSING AND NOISE REDUCTION 审中-公开
    用于音频噪声处理和噪声减少的系统和方法

    公开(公告)号:US20150325251A1

    公开(公告)日:2015-11-12

    申请号:US14274544

    申请日:2014-05-09

    Applicant: Apple Inc.

    CPC classification number: G10L21/0216 G10L21/0208 G10L2021/02166

    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.

    Abstract translation: 用于音频噪声处理和降噪的电子系统包括:第一和第二噪声估计器,选择器和衰减器。 第一噪声估计器处理来自语音波束形成器(VB)的第一音频信号并产生第一噪声估计。 VB通过来自第一和第二音频拾取通道的波束成形音频信号产生第一音频信号。 第二噪声估计器与第一噪声估计器并行地处理来自噪声波束形成器(NB)的第一和第二音频信号,并产生第二噪声估计。 NB通过波束成形来自第一和第二音频拾取通道的音频信号产生第二音频信号。 第一和第二音频信号包括第一和第二频率区域中的频率。 选择器的输出噪声估计可以是a)第一频率区域中的第二噪声估计,以及b)第二频率区域中的第一噪声估计。 衰减器根据输出噪声估计衰减第一音频信号。 还描述了其它实施例。

    NOISE ESTIMATION IN A MOBILE DEVICE USING AN EXTERNAL ACOUSTIC MICROPHONE SIGNAL
    52.
    发明申请
    NOISE ESTIMATION IN A MOBILE DEVICE USING AN EXTERNAL ACOUSTIC MICROPHONE SIGNAL 有权
    使用外部声音麦克风​​信号的移动设备中的噪声估计

    公开(公告)号:US20150296294A1

    公开(公告)日:2015-10-15

    申请号:US14248834

    申请日:2014-04-09

    Applicant: Apple Inc.

    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。 为了改善由内部麦克风产生的第一信号中的语音质量,移动设备识别出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。

    Adaptive Audio Feedback System and Method
    53.
    发明申请
    Adaptive Audio Feedback System and Method 审中-公开
    自适应音频反馈系统和方法

    公开(公告)号:US20130159861A1

    公开(公告)日:2013-06-20

    申请号:US13769217

    申请日:2013-02-15

    Applicant: Apple Inc.

    CPC classification number: G06F3/167 G06F3/0482 G10L13/00 G10L15/00

    Abstract: Various techniques for adaptively varying audio feedback data on an electronic device are provided. In one embodiment, an audio user interface implementing certain aspects of the present disclosure may devolve or evolve the verbosity of audio feedback in response to user interface events based at least partially upon the verbosity level of audio feedback provided during previous occurrences of the user interface event. In another embodiment, an audio user interface may be configured to vary the verbosity of audio feedback associated with a navigable list of items based at least partially upon the speed at which a user navigates the list. In a further embodiment, an audio user interface may be configured to vary audio feedback verbosity based upon the contextual importance of a user interface event. Electronic devices implementing the present techniques provide an improved user experience with regard to audio user interfaces.

    Abstract translation: 提供了用于在电子设备上自适应地改变音频反馈数据的各种技术。 在一个实施例中,实现本公开的某些方面的音频用户界面可以至少部分地基于在先前出现的用户界面事件期间提供的音频反馈的详细度级别来响应于用户界面事件来排放或演变音频反馈的冗长度 。 在另一个实施例中,音频用户界面可以被配置为至少部分地基于用户浏览列表的速度来改变与可导航的项目列表相关联的音频反馈的冗长度。 在另一个实施例中,音频用户界面可以被配置为基于用户界面事件的上下文重要性来改变音频反馈冗长度。 实现本技术的电子设备提供了关于音频用户界面的改进的用户体验。

    System and method for maintaining accuracy of voice recognition

    公开(公告)号:US10674303B2

    公开(公告)日:2020-06-02

    申请号:US16144851

    申请日:2018-09-27

    Applicant: Apple Inc.

    Inventor: Aram M. Lindahl

    Abstract: Method and system for maintaining accuracy of voice recognition are described herein. The audio system reproducing sound using a loudspeaker array that is housed in a loudspeaker cabinet may selection from a number of sound rendering modes and changing the selected sound rendering mode based on the current playback volume set on the audio system. The sound rendering modes include at least one of: a number of free space modes and a number of complex modes. Other aspects are also described and claimed.

    Method to determine loudspeaker change of placement

    公开(公告)号:US10567901B2

    公开(公告)日:2020-02-18

    申请号:US15514455

    申请日:2015-09-29

    Applicant: Apple Inc.

    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.

    Automatic speech recognition triggering system

    公开(公告)号:US10313782B2

    公开(公告)日:2019-06-04

    申请号:US15587325

    申请日:2017-05-04

    Applicant: Apple Inc.

    Abstract: An automatic speech recognition (ASR) triggering system, and a method of providing an ASR trigger signal, is described. The ASR triggering system can include a microphone to generate an acoustic signal representing an acoustic vibration and an accelerometer worn in an ear canal of a user to generate a non-acoustic signal representing a bone conduction vibration. A processor of the ASR triggering system can receive an acoustic trigger signal based on the acoustic signal and a non-acoustic trigger signal based on the non-acoustic signal, and combine the trigger signals to gate an ASR trigger signal. For example, the ASR trigger signal may be provided to an ASR server only when the trigger signals are simultaneously asserted. Other embodiments are also described and claimed.

    Multi-microphone speech recognition systems and related techniques

    公开(公告)号:US10304462B2

    公开(公告)日:2019-05-28

    申请号:US15871836

    申请日:2018-01-15

    Applicant: Apple Inc.

    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.

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