公开(公告)号：US20130293558A1

公开(公告)日：2013-11-07

申请号：US13875215

申请日：2013-05-01

Applicant: APPLE INC.

Inventor： Brett D. George ,  Changki Min ,  David A. Leech ,  Matthew X. Mora ,  Neil D. Warren ,  Rajabali M. Koduri ,  Ronald N. Isaac

IPC: G06T1/20

CPC classification number: G06T1/20 ,  G10K2210/3042

Abstract: Methods and apparatus for processing media signals. In one embodiment, a data processing device processes fixed and variable rate data using a first and second processing unit. The processing comprises real-time processing of audio/video signals by a graphics processing unit (GPU) and/or central processing unit (CPU). The processing units process data efficiently by establishing one processor as always processing variable rate data, and using one or more schemes for determining processor will process fixed rate data. A shared memory enables the processors to communicate with one another in order to determine which will process the fixed rate data. In one scheme for determining which of the processors will process the fixed rate data the second processor need merely be unlocked. In another embodiment, the second processor must be unlocked and immediately available.

Abstract translation: 用于处理媒体信号的方法和装置。 在一个实施例中，数据处理设备使用第一和第二处理单元处理固定和可变速率数据。 该处理包括由图形处理单元（GPU）和/或中央处理单元（CPU）对音频/视频信号的实时处理。 处理单元通过一如既往处理可变速率数据建立一个处理器，并且使用一个或多个用于确定处理器的方案将处理固定速率数据来有效地处理数据。 共享存储器使处理器能够彼此通信，以便确定哪个处理固定速率数据。 在用于确定哪些处理器将处理固定速率数据的一种方案中，第二处理器仅需要解锁。 在另一个实施例中，第二处理器必须被解锁并立即可用。

公开(公告)号：US20170099458A1

公开(公告)日：2017-04-06

申请号：US15311824

申请日：2014-09-26

Applicant: APPLE INC.

Inventor： Martin E. Johnson ,  Tom-Davy William Jendrik Saux ,  Ronald N. Isaac ,  Sylvain Choisel ,  Afrooz Family

IPC: H04N7/15 ,  H04R3/12 ,  H04N7/14 ,  H04S7/00

CPC classification number: H04N7/15 ,  H04N7/141 ,  H04R3/12 ,  H04R27/00 ,  H04R2430/01 ,  H04S7/30 ,  H04S7/305 ,  H04S2400/13 ,  H04S2400/15

Abstract: A video conferencing system is described that includes a near-end and a far-end system. The near-end system records both audio and video of one or more users proximate to the near-end system. This recorded audio and video is transmitted to the far-end system through the data connection. The video stream and/or one or more settings of the recording camera are analyzed to determine the amount of a video frame occupied by the recorded user(s). The video conferencing system may directly analyze the video frames themselves and/or a zoom setting of the recording camera to determine a ratio or percentage of the video frame occupied by the recorded user(s). By analyzing video frames associated with an audio stream, the video conferencing system may drive a speaker array of the far-end system to more accurately reproduce sound content based on the position of the recorded user in a video frame.

公开(公告)号：US09538308B2

公开(公告)日：2017-01-03

申请号：US14775585

申请日：2014-03-13

Applicant: Apple Inc.

Inventor： Ronald N. Isaac

IPC: H04R5/02 ,  H04S7/00

CPC classification number: H04S7/301 ,  H04S7/307 ,  H04S2400/01 ,  H04S2400/15 ,  H04S2420/07

Abstract: A loudspeaker that measures the impulse response of a listening area is described. The loudspeaker may output sounds corresponding to a segment of an audio signal. The sounds are sensed by a listening device proximate to a listener and transmitted to the loudspeaker. The loudspeaker includes an adaptive filter that estimates the impulse response of the listening area based on the signal segment. An error unit analyzes the estimated impulse response together with the sensed audio signal received from the listening device to determine the accuracy of the estimate. New estimates may be generated by the adaptive filter until an accuracy level is achieved for the signal segment. A processor may utilize one or more estimated impulse responses corresponding to various signal segments that cover a defined frequency spectrum for adjusting the audio signal to compensate for the impulse response of the listening area. Other embodiments are also described.

Abstract translation: 描述了测量收听区域的脉冲响应的扬声器。 扬声器可以输出与音频信号的段对应的声音。 声音由靠近收听者的聆听装置感测并被发送到扬声器。 该扬声器包括一个自适应滤波器，该自适应滤波器基于信号段估计收听区域的脉冲响应。 误差单元将所估计的脉冲响应与从听音装置接收到的感测音频信号一起分析，以确定估计的准确性。 可以通过自适应滤波器生成新的估计，直到信号段达到精度水平为止。 处理器可以利用对应于覆盖所定义的频谱的各种信号段的一个或多个估计的脉冲响应来调整音频信号以补偿收听区域的脉冲响应。 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。