AUDIO SPATIALIZATION USING REFLECTIVE ROOM MODEL
    1.
    发明申请
    AUDIO SPATIALIZATION USING REFLECTIVE ROOM MODEL 有权
    使用反射室模型的音频空间化

    公开(公告)号:US20110268281A1

    公开(公告)日:2011-11-03

    申请号:US12772014

    申请日:2010-04-30

    IPC分类号: H04R5/00

    摘要: Described are systems and methods performed by computer to reduce crosstalk produced by loudspeakers when rendering binaural sound that is emitted from the loudspeakers into a room. The room may have sound-reflecting surfaces that reflect some of the sound produced by the loudspeakers. To reduce crosstalk, a room model stored by the computer, is accessed. The room model models at least sound reflected by one or more of the physical surfaces. The room model is used to calculate a model of an audio channel from the loudspeakers to a listener. The model of the audio channel models sound transmission from the loudspeakers to the listener. The computer uses the model of the audio channel to cancel crosstalk from the loudspeakers when rendering the binaural sound.

    摘要翻译: 描述了由计算机执行的系统和方法,以便在将从扬声器发射到室内的双耳声音降低时,减少扬声器产生的串扰。 房间可能具有反映由扬声器产生的一些声音的声音反射表面。 为了减少串扰,计算机存储的房间模型被访问。 房间模型至少模拟一个或多个物理表面反射的声音。 房间模型用于计算从扬声器到收听者的音频通道的模型。 音频通道的模型模型从扬声器传输到听众。 当渲染双耳声音时,计算机使用音频通道的模型来消除扬声器的串扰。

    Audio spatialization using reflective room model
    2.
    发明授权
    Audio spatialization using reflective room model 有权
    使用反射室模型的音频空间化

    公开(公告)号:US09107021B2

    公开(公告)日:2015-08-11

    申请号:US12772014

    申请日:2010-04-30

    IPC分类号: H04S7/00 H04S1/00 G06F17/50

    摘要: Described are systems and methods performed by computer to reduce crosstalk produced by loudspeakers when rendering binaural sound that is emitted from the loudspeakers into a room. The room may have sound-reflecting surfaces that reflect some of the sound produced by the loudspeakers. To reduce crosstalk, a room model stored by the computer, is accessed. The room model models at least sound reflected by one or more of the physical surfaces. The room model is used to calculate a model of an audio channel from the loudspeakers to a listener. The model of the audio channel models sound transmission from the loudspeakers to the listener. The computer uses the model of the audio channel to cancel crosstalk from the loudspeakers when rendering the binaural sound.

    摘要翻译: 描述了由计算机执行的系统和方法,以便在将从扬声器发射到室内的双耳声音降低时,减少扬声器产生的串扰。 房间可能具有反映由扬声器产生的一些声音的声音反射表面。 为了减少串扰,计算机存储的房间模型被访问。 房间模型至少模拟一个或多个物理表面反射的声音。 房间模型用于计算从扬声器到收听者的音频通道的模型。 音频通道的模型模型从扬声器传输到听众。 当渲染双耳声音时,计算机使用音频通道的模型来消除扬声器的串扰。

    Immersive Remote Conferencing
    3.
    发明申请
    Immersive Remote Conferencing 有权
    沉浸式远程会议

    公开(公告)号:US20120281059A1

    公开(公告)日:2012-11-08

    申请号:US13100504

    申请日:2011-05-04

    IPC分类号: H04N7/15

    摘要: The subject disclosure is directed towards an immersive conference, in which participants in separate locations are brought together into a common virtual environment (scene), such that they appear to each other to be in a common space, with geometry, appearance, and real-time natural interaction (e.g., gestures) preserved. In one aspect, depth data and video data are processed to place remote participants in the common scene from the first person point of view of a local participant. Sound data may be spatially controlled, and parallax computed to provide a realistic experience. The scene may be augmented with various data, videos and other effects/animations.

    摘要翻译: 本发明涉及一种身临其境的会议,其中分开的位置的参与者被聚集到一个共同的虚拟环境(场景)中,使得它们彼此看起来处于共同的空间中,具有几何,外观和实时性, 保留时间自然的相互作用(如手势)。 在一个方面,深度数据和视频数据被处理以将远程参与者从本地参与者的第一人的角度放置在公共场景中。 声音数据可以是空间控制的,并且计算视差以提供真实的体验。 场景可能会增加各种数据,视频和其他效果/动画。

    Multi-view video compression and streaming based on viewpoints of remote viewer

    公开(公告)号:US09648346B2

    公开(公告)日:2017-05-09

    申请号:US12491775

    申请日:2009-06-25

    摘要: Multi-view video that is being streamed to a remote device in real time may be encoded. Frames of a real-world scene captured by respective video cameras are received for compression. A virtual viewpoint, positioned relative to the video cameras, is used to determine expected contributions of individual portions of the frames to a synthesized image of the scene from the viewpoint position using the frames. For each frame, compression rates for individual blocks of a frame are computed based on the determined contributions of the individual portions of the frame. The frames are compressed by compressing the blocks of the frames according to their respective determined compression rates. The frames are transmitted in compressed form via a network to a remote device, which is configured to render the scene using the compressed frames.

    Immersive remote conferencing
    5.
    发明授权
    Immersive remote conferencing 有权
    沉浸式远程会议

    公开(公告)号:US08675067B2

    公开(公告)日:2014-03-18

    申请号:US13100504

    申请日:2011-05-04

    IPC分类号: H04N7/18

    摘要: The subject disclosure is directed towards an immersive conference, in which participants in separate locations are brought together into a common virtual environment (scene), such that they appear to each other to be in a common space, with geometry, appearance, and real-time natural interaction (e.g., gestures) preserved. In one aspect, depth data and video data are processed to place remote participants in the common scene from the first person point of view of a local participant. Sound data may be spatially controlled, and parallax computed to provide a realistic experience. The scene may be augmented with various data, videos and other effects/animations.

    摘要翻译: 本发明涉及一种身临其境的会议,其中分开的位置的参与者被聚集到一个共同的虚拟环境(场景)中,使得它们彼此看起来处于共同的空间中,具有几何,外观和实时性, 保留时间自然的相互作用(如手势)。 在一个方面,深度数据和视频数据被处理以将远程参与者从本地参与者的第一人的角度放置在公共场景中。 声音数据可以是空间控制的,并且计算视差以提供真实的体验。 场景可能会增加各种数据,视频和其他效果/动画。

    Enhanced Beamforming for Arrays of Directional Microphones
    6.
    发明申请
    Enhanced Beamforming for Arrays of Directional Microphones 有权
    定向麦克风阵列的增强波束形成

    公开(公告)号:US20080240463A1

    公开(公告)日:2008-10-02

    申请号:US11692920

    申请日:2007-03-29

    IPC分类号: H04R3/10

    CPC分类号: H04R3/005

    摘要: A novel enhanced beamforming technique that improves beamforming operations by incorporating a model for the directional gains of the sensors, such as microphones, and provides means of estimating these gains. The technique forms estimates of the relative magnitude responses of the sensors (e.g., microphones) based on the data received at the array and includes those in the beamforming computations.

    摘要翻译: 一种新颖的增强波束形成技术,其通过结合用于诸如麦克风的传感器的定向增益的模型来改进波束成形操作,并且提供估计这些增益的手段。 该技术基于在阵列处接收到的数据并且包括波束成形计算中的数据来形成传感器(例如,麦克风)的相对幅度响应的估计。

    Entering confidential information on an untrusted machine
    7.
    发明授权
    Entering confidential information on an untrusted machine 有权
    在不受信任的机器上输入机密信息

    公开(公告)号:US08825728B2

    公开(公告)日:2014-09-02

    申请号:US11453626

    申请日:2006-06-15

    CPC分类号: H04L63/145 H04L63/083

    摘要: Confidential information is provided to a proxy computer in communication between an unsecured computer and a computer having information desired by a user. The proxy computer receives the confidential information in either an encrypted form or having arbitrary information combined therewith. The proxy computer ascertains the confidential information and forwards it to the computer having the information desired by the user.

    摘要翻译: 机密信息被提供给在不安全的计算机和具有用户期望的信息的计算机之间的通信中的代理计算机。 代理计算机以加密形式接收机密信息或者具有与之组合的任意信息。 代理计算机确定机密信息并将其转发给具有用户期望的信息的计算机。

    AUDIO TRANSFORMS IN CONNECTION WITH MULTIPARTY COMMUNICATION
    8.
    发明申请
    AUDIO TRANSFORMS IN CONNECTION WITH MULTIPARTY COMMUNICATION 有权
    与多媒体通信相关的音频转换

    公开(公告)号:US20100195812A1

    公开(公告)日:2010-08-05

    申请号:US12365949

    申请日:2009-02-05

    IPC分类号: H04M3/42 G10L11/00

    摘要: The claimed subject matter relates to an architecture that can preprocess audio portions of communications in order to enrich multiparty communication sessions or environments. In particular, the architecture can provide both a public channel for public communications that are received by substantially all connected parties and can further provide a private channel for private communications that are received by a selected subset of all connected parties. Most particularly, the architecture can apply an audio transform to communications that occur during the multiparty communication session based upon a target audience of the communication. By way of illustration, the architecture can apply a whisper transform to private communications, an emotion transform based upon relationships, an ambience or spatial transform based upon physical locations, or a pace transform based upon lack of presence.

    摘要翻译: 所要求保护的主题涉及可以预处理通信的音频部分以便丰富多方通信会话或环境的架构。 特别地,该架构可以提供公共通信的公共信道,其由基本上所有连接的各方接收,并且可以进一步提供由所有连接方的所选子集接收的专用通信的专用信道。 特别地,架构可以基于通信的目标受众对音频转换应用于在多方通信会话期间发生的通信。 作为说明,架构可以对私人通信应用耳语转换,基于关系,基于物理位置的氛围或空间变换或基于缺乏存在的步调变换的情感变换。

    REAL-TIME DETECTION AND PRESERVATION OF SPEECH ONSET IN A SIGNAL
    9.
    发明申请
    REAL-TIME DETECTION AND PRESERVATION OF SPEECH ONSET IN A SIGNAL 有权
    实时检测和保留信号中的声音

    公开(公告)号:US20080281586A1

    公开(公告)日:2008-11-13

    申请号:US12181159

    申请日:2008-07-28

    IPC分类号: G10L11/06

    CPC分类号: G10L25/87 G10L2025/783

    摘要: A “speech onset detector” provides a variable length frame buffer in combination with either variable transmission rate or temporal speech compression for buffered signal frames. The variable length buffer buffers frames that are not clearly identified as either speech or non-speech frames during an initial analysis. Buffering of signal frames continues until a current frame is identified as either speech or non-speech. If the current frame is identified as non-speech, buffered frames are encoded as non-speech frames. However, if the current frame is identified as a speech frame, buffered frames are searched for the actual onset point of the speech. Once that onset point is identified, the signal is either transmitted in a burst, or a time-scale modification of the buffered signal is applied for compressing buffered frames beginning with the frame in which onset point is detected. The compressed frames are then encoded as one or more speech frames.

    摘要翻译: “语音起始检测器”提供了可变长度帧缓冲器,与缓冲信号帧的可变传输速率或时间语音压缩相结合。 可变长度缓冲器缓冲在初始分析期间未被清楚地识别为语音或非语音帧的帧。 信号帧的缓冲持续到当前帧被识别为语音或非语音。 如果当前帧被识别为非语音,则缓冲帧被编码为非语音帧。 然而,如果当前帧被识别为语音帧,则搜索缓冲的帧用于语音的实际起始点。 一旦该起始点被识别,则信号以突发方式发送,或者缓冲信号的时间尺度修改被应用于从检测到起始点的帧开始的缓冲帧。 然后将压缩的帧编码为一个或多个语音帧。