公开(公告)号：US09721984B2

公开(公告)日：2017-08-01

申请号：US13445766

申请日：2012-04-12

Applicant: Mu-Han Cheng ,  Kuo-Hsiu Wei ,  Kei-Wei Chen ,  Ying-Lang Wang

Inventor： Mu-Han Cheng ,  Kuo-Hsiu Wei ,  Kei-Wei Chen ,  Ying-Lang Wang

IPC: H01L27/146

CPC classification number: H01L27/14623 ,  H01L27/1464 ,  H01L27/14687

Abstract: Semiconductor devices and back side illumination (BSI) sensor manufacturing methods are disclosed. In one embodiment, a method of manufacturing a semiconductor device includes providing a workpiece and forming an integrated circuit on a front side of the workpiece. A grid of a conductive material is formed on a back side of the workpiece using a damascene process.

公开(公告)号：US20130273686A1

公开(公告)日：2013-10-17

申请号：US13445766

申请日：2012-04-12

Applicant: Mu-Han Cheng ,  Kuo-Hsiu Wei ,  Kei-Wei Chen ,  Ying-Lang Wang

Inventor： Mu-Han Cheng ,  Kuo-Hsiu Wei ,  Kei-Wei Chen ,  Ying-Lang Wang

IPC: H01L31/0232 ,  H01L21/28

CPC classification number: H01L27/14623 ,  H01L27/1464 ,  H01L27/14687

Abstract: Semiconductor devices and back side illumination (BSI) sensor manufacturing methods are disclosed. In one embodiment, a method of manufacturing a semiconductor device includes providing a workpiece and forming an integrated circuit on a front side of the workpiece. A grid of a conductive material is formed on a back side of the workpiece using a damascene process.

Abstract translation: 公开了半导体器件和背面照明（BSI）传感器制造方法。 在一个实施例中，制造半导体器件的方法包括提供工件并在工件的前侧形成集成电路。 使用镶嵌工艺在工件的背面上形成导电材料的格栅。

公开(公告)号：US08295475B2

公开(公告)日：2012-10-23

申请号：US11332500

申请日：2006-01-13

Applicant: Qin Li ,  Chao He ,  Wei-Ge Chen ,  Mu Han

Inventor： Qin Li ,  Chao He ,  Wei-Ge Chen ,  Mu Han

IPC: H04M9/08

CPC classification number: H04M9/082

Abstract: The quality and robustness of audio echo cancellation is enhanced by selectively applying glitch recovery processes based on a quality measurement of the relative offset between capture and render audio streams. For example, large and small glitch detection is enabled for low relative offset variance; large glitch detection is enabled in a medium range of relative offset variance; and neither enabled at high variance. Further, a fast glitch recovery process suspends updating the adaptive filter coefficients of the audio echo cancellation while buffers are re-aligned to recover from the glitch, so as to avoid resetting the adaptive filter. When clock drift exists between capture and render audio streams, a multi-step compensation method is applied to improve AEC output quality in case the drifting rate is low; and a resampler is used to compensate the drift in case the drifting rate is high. An anti-clipping process detects clipping of the signals, and also suspends adaptive filter updating during clipping.

Abstract translation: 通过基于捕获和渲染音频流之间的相对偏移的质量测量来选择性地应用毛刺恢复过程来增强音频回声消除的质量和鲁棒性。 例如，对于低相对偏移方差，启用大的和小的毛刺检测; 在相对偏移方差的中等范围内启用大毛刺检测; 并且在高方差时都不启用。 此外，快速毛刺恢复处理暂停更新音频回声消除的自适应滤波器系数，同时缓冲器被重新对准以从毛刺恢复，以避免复位自适应滤波器。 当捕获和渲染音频流之间存在时钟漂移时，应用多步补偿方法来提高漂移率低的AEC输出质量; 并且在漂移速率高的情况下使用重采样器来补偿漂移。 反剪辑过程检测信号的剪辑，并且还可以在剪辑期间暂停自适应滤波器更新。

公开(公告)号：US20100149301A1

公开(公告)日：2010-06-17

申请号：US12334836

申请日：2008-12-15

Applicant: Ming-Chieh Lee ,  Mu Han ,  Tim Moore

Inventor： Ming-Chieh Lee ,  Mu Han ,  Tim Moore

IPC: H04N7/14

CPC classification number: H04N7/15 ,  H04L12/1818 ,  H04L12/1877 ,  H04N7/147 ,  H04N21/42203 ,  H04N21/4223 ,  H04N21/440263 ,  H04N21/440281 ,  H04N21/4788 ,  H04N21/6332 ,  H04N21/6377 ,  H04N21/658

Abstract: Subscriptions in a video conference may be provided using multiple bit rate streams. A video conference server may receive video streams from each client in a video conference and may receive subscription requests from each client. The subscription requests may include requests to see video streams from specific other clients at a given resolution and/or frame rate. The video conference server may match up the received video streams with the subscription requests in order to send the subscribing clients their desired video streams. The server may also be able to request different versions of video streams from participants (e.g. different resolutions) and/or alter the video streams in order to better comply with the subscription request.

Abstract translation: 可以使用多个比特率流来提供视频会议中的订阅。 视频会议服务器可以在视频会议中从每个客户端接收视频流，并且可以接收来自每个客户端的订阅请求。 订阅请求可以包括以给定分辨率和/或帧速率查看来自特定其他客户端的视频流的请求。 视频会议服务器可以将接收到的视频流与订阅请求相匹配，以便向订阅的客户端发送其期望的视频流。 服务器还可以从参与者请求不同版本的视频流（例如，不同的分辨率）和/或改变视频流，以便更好地符合订阅请求。

公开(公告)号：US07554922B2

公开(公告)日：2009-06-30

申请号：US11560445

申请日：2006-11-16

Applicant: Andres Vega-Garcia ,  Mu Han ,  Qianbo Huai

Inventor： Andres Vega-Garcia ,  Mu Han ,  Qianbo Huai

IPC: H04J3/14

CPC classification number: H04L65/608 ,  H04L29/06 ,  H04L29/06027 ,  H04L47/10 ,  H04L47/115 ,  H04L47/263 ,  H04L47/283 ,  H04L65/80

Abstract: A method and system for dynamically altering the transmission settings of one or more computing devices engaged in a real-time communication session is presented. The devices exchange meaningful and dummy control packets according to a standard control protocol. The approximate bandwidth available on the network is then calculated based on the difference in arrival times between at least one of the dummy control packets and at least one of the meaningful control packets. Once the approximate bandwidth available on the network is computed, the one or more devices adjust outgoing audio and video data streams using a quality control mechanism. The quality control mechanism enables the one or more devices to transmit data in a way that maximizes the user experience during the real-time communication session.

Abstract translation: 提出了一种用于动态地改变参与实时通信会话的一个或多个计算设备的传输设置的方法和系统。 设备根据标准控制协议交换有意义的和虚拟的控制数据包。 然后基于至少一个虚拟控制分组与至少一个有意义的控制分组之间的到达时间的差异来计算网络上可用的大致带宽。 一旦计算了网络上可用的大致带宽，则一个或多个设备使用质量控制机制来调整输出的音频和视频数据流。 质量控制机制使得一个或多个设备能够以在实时通信会话期间最大化用户体验的方式来发送数据。

公开(公告)号：US20080294793A1

公开(公告)日：2008-11-27

申请号：US11951912

申请日：2007-12-06

Applicant: Mu Han ,  Andres Vega Garcia ,  Wei Zhong

Inventor： Mu Han ,  Andres Vega Garcia ,  Wei Zhong

IPC: G06F15/16

CPC classification number: H04L12/6418 ,  H04L2012/6472 ,  Y10S345/951

Abstract: A technique for reducing information reception delays is provided. The technique reduces delays that may be caused by protocols that guarantee order and delivery, such as TCP/IP. The technique creates multiple connections between a sender and recipient computing devices and sends messages from the sender to the recipient on the multiple corrections redundantly. The recipient can then use the first arriving message and ignore the subsequently arriving redundant messages. The recipient can also wait for a period of time before determining which of the arrived messages to use. The technique may dynamically add connections if messages are not consistently received in a timely manner on multiple connections. Conversely, the technique may remove connections if messages are consistently received in a timely manner on multiple connections. The technique can accordingly be used with applications that are intolerant of data reception delays such as Voice over IP, real-time streaming audio, or real-time streaming video.

Abstract translation: 提供了用于减少信息接收延迟的技术。 该技术减少了可能由保证订单和传递的协议（如TCP / IP）引起的延迟。 该技术在发送方和收件人计算设备之间创建多个连接，并以多次更正方式从发送方向接收方发送消息。 接收者可以使用第一个到达的消息，并忽略随后到达的冗余消息。 收件人还可以等待一段时间才能确定要使用的到达消息。 如果在多个连接上不及时地接收到消息，则该技术可以动态地添加连接。 相反，如果在多个连接上一致地接收到消息，则该技术可以去除连接。 因此，该技术可以与不耐受诸如IP语音，实时流音频或实时流视频之类的数据接收延迟的应用一起使用。

公开(公告)号：US08380790B2

公开(公告)日：2013-02-19

申请号：US12334969

申请日：2008-12-15

Applicant: Ming Chieh Lee ,  Mu Han ,  Tim Moore

Inventor： Ming Chieh Lee ,  Mu Han ,  Tim Moore

IPC: G06F13/00

CPC classification number: H04N7/15 ,  H04L12/1827 ,  H04N21/25833 ,  H04N21/44029 ,  H04N21/4788 ,  H04N21/6377 ,  H04N21/658

Abstract: Video conference rate matching may be provided. A video conference server may receive video source streams from clients on a video conference. The server may analyze each client's capabilities and choose a video stream to send to each client based on those capabilities. For example, a client capable of encoding and decoding a high definition video stream may provide three source video streams—a high definition stream, a medium resolution stream, and a low resolution stream. The server may send only the low resolution stream to a client with a low amount of available bandwidth. The server may send the medium resolution stream to another client with sufficient bandwidth for the high definition stream, but which lacks the ability to decode the high definition stream.

Abstract translation: 可以提供视频会议速率匹配。 视频会议服务器可以在视频会议上从客户接收视频源流。 服务器可以分析每个客户端的功能，并根据这些功能选择一个视频流来发送给每个客户端。 例如，能够对高分辨率视频流进行编码和解码的客户端可以提供三个源视频流 - 高清晰度流，中分辨率流和低分辨率流。 服务器只能将低分辨率流发送到具有低可用带​​宽的客户端。 服务器可以向具有足够带宽的高清晰度流的另一个客户端发送媒体分辨率流，但是缺少解码高分辨率流的能力。

公开(公告)号：US08321506B2

公开(公告)日：2012-11-27

申请号：US10918855

申请日：2004-08-14

Applicant: Mu Han ,  Krishnamurthy Ganesan ,  Adrian Potra ,  Nikhil Bobde

Inventor： Mu Han ,  Krishnamurthy Ganesan ,  Adrian Potra ,  Nikhil Bobde

IPC: G06F15/16 ,  H04Q11/00

CPC classification number: H04L65/1006 ,  G06F9/54 ,  H04L29/06 ,  H04L29/06027 ,  H04L65/403

Abstract: An architecture for an extensible real-time collaboration system is provided. The architecture presents a unified application program interface for writing application programs that use communications protocols. The architecture has activity objects, endpoint objects, and multiple media stacks. These objects may use various communications protocols, such as Session Initiation Protocol or Real-Time Transport Protocol to send and receive messages. The activity objects, endpoint objects, and multiple media stacks may each have one or more APIs that an application developer can use to access or provide collaboration-related functionality. These objects map the API to the underlying implementation provided by other objects. Using the activity objects enables a developer to provide less application logic than would otherwise be necessary to provide complex collaboration services.

Abstract translation: 提供了可扩展实时协作系统的架构。 该架构提供了一个统一的应用程序界面，用于编写使用通信协议的应用程序。 该架构具有活动对象，端点对象和多个媒体堆栈。 这些对象可以使用各种通信协议，例如会话发起协议或实时传输协议来发送和接收消息。 活动对象，端点对象和多个媒体堆栈可以各自具有应用开发者可以用来访问或提供协作相关功能的一个或多个API。 这些对象将API映射到其他对象提供的底层实现。 使用活动对象使开发人员能够提供比提供复杂协作服务所必需的更少的应用程序逻辑。

公开(公告)号：US07747801B2

公开(公告)日：2010-06-29

申请号：US11951912

申请日：2007-12-06

Applicant: Mu Han ,  Andres Vega Garcia ,  Wei Zhong

Inventor： Mu Han ,  Andres Vega Garcia ,  Wei Zhong

IPC: H04L12/28 ,  G01R31/08

CPC classification number: H04L12/6418 ,  H04L2012/6472 ,  Y10S345/951

Abstract: A technique for reducing information reception delays is provided. The technique reduces delays that may be caused by protocols that guarantee order and delivery, such as TCP/IP. The technique creates multiple connections between a sender and recipient computing devices and sends messages from the sender to the recipient on the multiple corrections redundantly. The recipient can then use the first arriving message and ignore the subsequently arriving redundant messages. The recipient can also wait for a period of time before determining which of the arrived messages to use. The technique may dynamically add connections if messages are not consistently received in a timely manner on multiple connections. Conversely, the technique may remove connections if messages are consistently received in a timely manner on multiple connections. The technique can accordingly be used with applications that are intolerant of data reception delays such as Voice over IP, real-time streaming audio, or real-time streaming video.

Abstract translation: 提供了用于减少信息接收延迟的技术。 该技术减少了可能由保证订单和传递的协议（如TCP / IP）引起的延迟。 该技术在发送方和收件人计算设备之间创建多个连接，并以多次更正方式从发送方向接收方发送消息。 接收者可以使用第一个到达的消息，并忽略随后到达的冗余消息。 收件人还可以等待一段时间才能确定要使用的到达消息。 如果在多个连接上不及时地接收到消息，则该技术可以动态地添加连接。 相反，如果在多个连接上一致地接收到消息，则该技术可以去除连接。 因此，该技术可以与不耐受诸如IP语音，实时流音频或实时流视频之类的数据接收延迟的应用一起使用。

公开(公告)号：US20100125455A1

公开(公告)日：2010-05-20

申请号：US12692417

申请日：2010-01-22

Applicant: Tian Wang ,  Hosam A. Khalil ,  Kazuhito Koishida ,  Wei-Ge Chen ,  Mu Han

Inventor： Tian Wang ,  Hosam A. Khalil ,  Kazuhito Koishida ,  Wei-Ge Chen ,  Mu Han

IPC: G10L19/08

CPC classification number: G10L19/08 ,  G10L19/005 ,  G10L19/22

Abstract: Various strategies for rate/quality control and loss resiliency in an audio codec are described. The various strategies can be used in combination or independently. For example, a real-time speech codec uses intra frame coding/decoding, adaptive multi-mode forward error correction [“FEC”], and rate/quality control techniques. Intra frames help a decoder recover quickly from packet losses, while compression efficiency is still emphasized with predicted frames. Various strategies for inserting intra frames and signaling intra/predicted frames are described. With the adaptive multi-mode FEC, an encoder adaptively selects between multiple modes to efficiently and quickly provide a level of FEC that takes into account the bandwidth currently available for FEC. The FEC information itself may be predictively encoded and decoded relative to primary encoded information. Various rate/quality and FEC control strategies allow additional adaptation to available bandwidth and network conditions.

Abstract translation: 描述了音频编解码器中的速率/质量控制和丢失弹性的各种策略。 各种策略可以组合使用或独立使用。 例如，实时语音编解码器使用帧内编码/解码，自适应多模式前向纠错[“FEC”]和速率/质量控制技术。 帧内帧帮助解码器从分组丢失中快速恢复，而预测帧仍然强调压缩效率。 描述了用于插入帧内和信令帧内/预测帧的各种策略。 利用自适应多模式FEC，编码器在多种模式之间自适应地选择以有效且快速地提供考虑到当前可用于FEC的带宽的FEC级别。 FEC信息本身可以相对于主编码信息进行预测编码和解码。 各种速率/质量和FEC控制策略允许对可用带宽和网络条件进行额外的调整。