公开(公告)号：US09525392B2

公开(公告)日：2016-12-20

申请号：US14602001

申请日：2015-01-21

Applicant: Apple Inc.

Inventor： Arvindh Krishnaswamy

IPC: H03G3/02

CPC classification number: H03G3/02 ,  H03G3/3005 ,  H03G3/32 ,  H03G9/025

Abstract: Method of dynamically adapting playback volume on electronic device starts with processor receiving first user input and first portion of audio content. First user input signals to device to increase or decrease volume of sound output. Processor determines first loudness metric corresponding to first portion of audio content when first user input is received. First loudness metric is measure of loudness of first portion of audio content being outputted. Processor then stores in memory first loudness metric in association with first user input. Memory stores history of loudness metrics in association with user inputs. Processor then determines second loudness metric that is measure of loudness of second portion of audio content that is received and determines second user input associated with second loudness metric using history. Processor generates control signal to automatically control volume of sound output by device corresponding to second user input. Other embodiments are also described.

Abstract translation: 在电子设备上动态地适应播放音量的方法从处理器接收第一用户输入和音频内容的第一部分开始。 第一个用户输入信号到设备，以增加或减少声音输出的音量。 当接收到第一用户输入时，处理器确定与音频内容的第一部分对应的第一响度度量。 第一响度度量是输出的音频内容的第一部分的响度的度量。 然后处理器与第一用户输入相关联地存储在存储器中的第一响度度量。 内存存储与用户输入相关联的响度度量的历史。 然后处理器确定作为接收的音频内容的第二部分的响度的度量的第二响度度量，并且确定与使用历史的第二响度度量相关联的第二用户输入。 处理器产生控制信号，以自动控制与第二用户输入相对应的设备输出的音量。 还描述了其它实施例。

公开(公告)号：US20160358606A1

公开(公告)日：2016-12-08

申请号：US14732711

申请日：2015-06-06

Applicant: Apple Inc.

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

IPC: G10L15/26 ,  G10L15/02 ,  G10L15/06 ,  G10L15/01

CPC classification number: G10L15/32 ,  G10L15/20

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.

公开(公告)号：US09438194B2

公开(公告)日：2016-09-06

申请号：US14502918

申请日：2014-09-30

Applicant: Apple Inc.

Inventor： Arvindh Krishnaswamy ,  Joseph M. Williams

IPC: H03G3/20 ,  H03G3/32

CPC classification number: H03G3/32

Abstract: Method of dynamically adapting user volume input range on mobile device having global volume range starts by receiving a volume input selection from a user that is level included in user volume input range. User volume input range is a portion of global volume range. Device's processor then detects ambient noise level surrounding device and adjusts user volume input range from current portion of global volume range to different portion of global volume range based on detected ambient noise level. Volume input selection remains at the same level included in user volume input range after user volume input range is adjusted. Processor may identify sound profile that corresponds to ambient noise level being detected and adjusts user volume input range to a different portion of the global volume range based on identified sound profile. Other embodiments are also described.

Abstract translation: 在具有全局音量范围的移动设备上动态调整用户音量输入范围的方法从通过从包括在用户音量输入范围中的级别的用户接收音量输入选择开始。 用户音量输入范围是全局音量范围的一部分。 然后，设备的处理器检测围绕设备的环境噪声水平，并根据检测到的环境噪声水平将用户音量输入范围从全局音量范围的当前部分调整到全局音量范围的不同部分。 用户音量输入范围调整后，音量输入选择保持在用户音量输入范围内的相同级别。 处理器可以识别对应于正被检测到的环境噪声水平的声音简档，并且基于所识别的声音简档将用户音量输入范围调整到全局音量范围的不同部分。 还描述了其它实施例。

公开(公告)号：US09401685B2

公开(公告)日：2016-07-26

申请号：US13800487

申请日：2013-03-13

Applicant: Apple Inc.

Inventor： Arvindh Krishnaswamy ,  Juha O. Merimaa ,  Kapil Krishnamurthy ,  Yuchao Song

IPC: H03G3/20 ,  H03G3/30

CPC classification number: H03G3/3005 ,  H03G3/20 ,  H03G3/3089

Abstract: Automatic gain control systems disclosed herein can incorporate a confidence metric that can estimate the accuracy of gain adjustments calculated by an automatic gain control module. The confidence metric may be based on a percentage of valid audio samples in a given period of time. Based on the confidence metric, the AGC response may be reduced, delayed, frozen, or otherwise altered from the baseline gain adjustment. Time-averaging process may be used to estimate the input signal power level and determine an appropriate baseline gain adjustment. Additionally, weighting functions can be adjusted to prevent overestimation of the signal power.

Abstract translation: 本文公开的自动增益控制系统可以包括可以估计由自动增益控制模块计算的增益调整的精度的置信度量度。 可信度度量可以基于给定时间段内的有效音频样本的百分比。 基于可信度度量，AGC响应可以被减小，延迟，冻结或者从基线增益调整改变。 时间平均过程可用于估计输入信号功率电平并确定适当的基线增益调整。 此外，可以调整加权函数以防止信号功率的过高估计。

公开(公告)号：US20150341008A1

公开(公告)日：2015-11-26

申请号：US14502997

申请日：2014-09-30

Applicant: Apple Inc.

Inventor： Arvindh Krishnaswamy ,  Joseph M. Williams

IPC: H03G5/16

CPC classification number: H03G5/165 ,  H03G5/005

Abstract: An equalizer that linearly interpolates between two equalization states when transitioning from one equalization state to the other equalization state is described. The equalizer includes a transfer function generator and an equalization module. Each equalization state is defined or determined based on a set of parameters. The transfer function generator generates a set of interpolated transfer functions by performing linear interpolation on a first equalization state and a second equalization state based on the set of parameters. The linear interpolation is performed on corresponding Z-domain poles and zeros of the transfer functions of the first and second equalization states. The equalization module applies the set of interpolated transfer functions generated by the transfer function generator to an input audio signal.

Abstract translation: 描述了当从一个均衡状态转换到另一个均衡状态时在两个均衡状态之间线性内插的均衡器。 均衡器包括传递函数发生器和均衡模块。 基于一组参数来定义或确定每个均衡状态。 传递函数发生器通过基于该参数集在第一均衡状态和第二均衡状态上执行线性插值来生成一组内插传递函数。 在第一和第二均衡状态的传递函数的相应的Z域极点和零点上执行线性内插。 均衡模块将由传递函数发生器产生的内插传递函数集合应用于输入音频信号。

公开(公告)号：US20130329912A1

公开(公告)日：2013-12-12

申请号：US13800487

申请日：2013-03-13

Applicant: APPLE INC.

Inventor： Arvindh Krishnaswamy ,  Juha O. Merimaa ,  Kapil Krishnamurthy ,  Yuchao Song

IPC: H03G3/20

CPC classification number: H03G3/3005 ,  H03G3/20 ,  H03G3/3089

Abstract: Automatic gain control systems disclosed herein can incorporate a confidence metric that can estimate the accuracy of gain adjustments calculated by an automatic gain control module. The confidence metric may be based on a percentage of valid audio samples in a given period of time. Based on the confidence metric, the AGC response may be reduced, delayed, frozen, or otherwise altered from the baseline gain adjustment. Time-averaging process may be used to estimate the input signal power level and determine an appropriate baseline gain adjustment. Additionally, weighting functions can be adjusted to prevent overestimation of the signal power.

Abstract translation: 本文公开的自动增益控制系统可以包括可以估计由自动增益控制模块计算的增益调整的精度的置信度量度。 可信度度量可以基于给定时间段内的有效音频样本的百分比。 基于可信度度量，AGC响应可以被减小，延迟，冻结或者从基线增益调整改变。 时间平均过程可用于估计输入信号功率电平并确定适当的基线增益调整。 此外，可以调整加权函数以防止信号功率的过高估计。

公开(公告)号：US11513661B2

公开(公告)日：2022-11-29

申请号：US16936164

申请日：2020-07-22

Applicant: Apple Inc.

Inventor： Roberto Garcia ,  Anil K. Kandangath ,  Arvindh Krishnaswamy ,  Xiaoyuan Tu ,  Justin Wood

IPC: G06F3/01 ,  G06F3/0482 ,  H04L51/046 ,  G06F3/0488 ,  H04L51/04 ,  H04L51/08 ,  G06F1/16 ,  G06Q10/10 ,  H04W4/02 ,  H04N5/232 ,  H04M1/72454 ,  H04L51/42 ,  H04L51/222 ,  H04L51/234 ,  G06F3/0481 ,  G06F3/041 ,  G06F3/04842 ,  H04L51/10 ,  G06F3/04817 ,  G06F3/0487 ,  G06F3/04883 ,  G06F3/16 ,  G06F3/0485 ,  G06F3/04886 ,  G06F3/0346

Abstract: An electronic device includes a touch-sensitive surface, a display, and a camera sensor. The device displays a message region for displaying a message conversation and receives a request to add media to the message conversation. Responsive to receiving the request, the device displays a media selection interface concurrently with at least a portion of the message conversation. The media selection interface includes a plurality of affordances for selecting media for addition to the message conversation, the plurality of affordances includes a live preview affordance, at least a subset of the plurality of affordances includes thumbnail representations of media available for adding to the message conversation, and the live preview affordance is associated with a live camera preview. Responsive to detecting selection of the live preview affordance, the device captures a new image based on the live camera preview and selects the new image for addition to the message conversation.

公开(公告)号：US10540984B1

公开(公告)日：2020-01-21

申请号：US15273568

申请日：2016-09-22

Applicant: Apple Inc.

Inventor： Sarmad Aziz Malik ,  Arvindh Krishnaswamy

IPC: G10L21/0232 ,  G10L21/0388 ,  G10L21/0208

Abstract: Method for echo control using adaptive polynomial filters in sub-band domain starts with loudspeaker that is configured to be driven by a reference signal outputting a loudspeaker signal. Microphone receives at least one of: a near-end speaker signal, ambient noise signal, or the loudspeaker signal and generates a microphone signal. Adaptive polynomial filters in sub-band domain included in adaptive echo canceller (AEC) are configured to adaptively filter representation of the reference signal in a plurality of channels in a sub-band domain based on a clean signal to generate the echo estimate. Echo suppressor is configured to remove an echo estimate from the microphone signal to generate the clean signal. Other embodiments are described.

公开(公告)号：US20190251974A1

公开(公告)日：2019-08-15

申请号：US16389697

申请日：2019-04-19

Applicant: Apple Inc.

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

IPC: G10L15/32 ,  G10L15/20

CPC classification number: G10L15/32 ,  G10L15/20

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.

公开(公告)号：US10013981B2

公开(公告)日：2018-07-03

申请号：US14732711

申请日：2015-06-06

Applicant: Apple Inc.

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

IPC: G10L15/26 ,  G10L15/06 ,  G10L15/32 ,  G10L15/20

CPC classification number: G10L15/32 ,  G10L15/20

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.