公开(公告)号：US07194084B2

公开(公告)日：2007-03-20

申请号：US11239542

申请日：2005-09-28

Applicant: Shmuel Shaffer ,  Michael E. Knappe

Inventor： Shmuel Shaffer ,  Michael E. Knappe

IPC: H04M9/08

CPC classification number: H04R27/00 ,  G10L19/008

Abstract: Systems and methods are disclosed for packet voice conferencing. An encoding system accepts two sound field signals, representing the same sound field sampled at two spatially-separated points. The relative delay between the two sound field signals is detected over a given time interval. The sound field signals are combined and then encoded as a single audio signal, e.g., by a method suitable for monophonic VoIP. The encoded audio payload and the relative delay are placed in one or more packets and sent to a decoding device via the packet network.The decoding device uses the relative delay to drive a playout splitter—once the encoded audio payload has been decoded, the playout splitter creates multiple presentation channels by inserting the transmitted relative delay in the decoded signal for one (or more) of the presentation channels. The listener thus perceives a speaker's voice as originating from a location related to the speaker's physical position at the other end of the conference. An advantage of these embodiments is that a pseudo-stereo conference can be conducted with virtually the same bandwidth as a monophonic conference.

Abstract translation: 公开了用于分组语音会议的系统和方法。 编码系统接收两个声场信号，表示在两个空间分离点采样的相同声场。 在给定的时间间隔内检测两个声场信号之间的相对延迟。 声场信号被组合，然后被编码为单个音频信号，例如通过适合于单声道VoIP的方法。 编码的音频有效载荷和相对延迟被放置在一个或多个分组中，并且经由分组网络被发送到解码装置。 解码装置使用相对延迟来驱动播出分离器 - 一旦编码的音频有效载荷已被解码，播放分离器通过将一个（或更多个）演示频道的解码信号中插入传输的相对延迟来创建多个显示频道。 因此，听众将演讲者的声音从与会议另一端的演讲人的身体位置相关的位置发出。 这些实施例的优点是可以以与单声道会议几乎相同的带宽进行伪立体会议。

公开(公告)号：US06850496B1

公开(公告)日：2005-02-01

申请号：US09591891

申请日：2000-06-09

Applicant: Michael E. Knappe ,  Shmuel Shaffer

Inventor： Michael E. Knappe ,  Shmuel Shaffer

IPC: H04M3/56 ,  H04M7/00 ,  H04L12/18

CPC classification number: H04L65/403 ,  H04L65/4038 ,  H04M3/56 ,  H04M3/568 ,  H04M7/006

Abstract: A system and method are disclosed for packet voice conferencing. The system and method divide a conferencing presentation sound field into sectors, and allocate one or more sectors to each conferencing endpoint. At some point between capture and playout, the voice data from each endpoint is mapped into its designated sector or sectors. Thereafter, when the voice data from a plurality of participants from multiple endpoints is combined, a listener can identify a unique apparent location within the presentation sound field for each participant. The system allows a conference participant to increase their comprehension when multiple participants speak simultaneously, as well as alleviate confusion as to who is speaking at any given time.

Abstract translation: 公开了用于分组语音会议的系统和方法。 系统和方法将会议演示声场分成扇区，并为每个会议端点分配一个或多个扇区。 在捕获和播放之间的某个时刻，来自每个端点的语音数据被映射到其指定的扇区或扇区中。 此后，当来自多个端点的多个参与者的语音数据被组合时，收听者可以为每个参与者识别呈现声场内的唯一表观位置。 该系统允许会议参与者在多个参与者同时进行演讲时增加理解能力，并减轻在任何时候发言者的混淆。

公开(公告)号：US06240084B1

公开(公告)日：2001-05-29

申请号：US08729245

申请日：1996-10-10

Applicant: David R. Oran ,  Cary W. FitzGerald ,  Michael E. Knappe

Inventor： David R. Oran ,  Cary W. FitzGerald ,  Michael E. Knappe

IPC: H04L1266

CPC classification number: H04L12/66

Abstract: A PC-based server platform includes a first backplane bus used for transferring data and commands to various PC peripheral devices. A network router and a telephony endpoint card are coupled to the backplane bus and separately coupled through a second Time Division Multiplexed (TDM) bus. The router includes interfaces to various packet switched networks such as a Wide Area Network (WAN) and a Local Area Network (LAN). The TDM bus is used to route telephony data between the different Internet Protocol (IP)-based networks and the telephony card independently of the host system. The PC host processor also uses the router as a standard LAN interface for transferring data packets. A DSP voice processing card is coupled between the backplane bus and the TDM bus to compress and decompress the telephony data transferred on the TDM bus.

Abstract translation: 基于PC的服务器平台包括用于将数据和命令传送到各种PC外围设备的第一个背板总线。 网络路由器和电话端点卡耦合到背板总线，并通过第二时分复用（TDM）总线单独耦合。 路由器包括各种分组交换网络的接口，例如广域网（WAN）和局域网（LAN）。 TDM总线用于在不同的基于互联网协议（IP）的网络和独立于主机系统的电话卡之间路由电话数据。 PC主机处理器还使用路由器作为传输数据包的标准LAN接口。 DSP语音处理卡耦合在背板总线和TDM总线之间，以压缩和解压缩TDM总线上传送的电话数据。

公开(公告)号：US07656816B2

公开(公告)日：2010-02-02

申请号：US11466597

申请日：2006-08-23

Applicant: Shmuel Shaffer ,  Joseph F. Khouri ,  Michael E. Knappe ,  Robert T. Bell

Inventor： Shmuel Shaffer ,  Joseph F. Khouri ,  Michael E. Knappe ,  Robert T. Bell

IPC: H04L12/26

CPC classification number: H04M7/006 ,  G10L25/69 ,  H04M3/2236 ,  H04M3/42221 ,  H04M7/123 ,  H04M2201/36

Abstract: A method and system for logging voice quality issues for a communication connection includes receiving a signal for logging quality information for a voice connection at an endpoint of the voice connection. Voice samples are collected from the voice connection at the endpoint. The voice samples are stored in an error log at the endpoint.

Abstract translation: 用于记录通信连接的语音质量问题的方法和系统包括：在语音连接的端点处接收用于记录用于语音连接的质量信息的信号。 语音采样从终端的语音连接收集。 语音样本存储在端点的错误日志中。

公开(公告)号：US07453828B1

公开(公告)日：2008-11-18

申请号：US11611080

申请日：2006-12-14

Applicant: Luke K. Surazski ,  Pascal Huart ,  Michael E. Knappe

Inventor： Luke K. Surazski ,  Pascal Huart ,  Michael E. Knappe

IPC: H04L12/16 ,  H04J3/16

CPC classification number: H04L29/06027 ,  H04L65/1083 ,  H04L65/403 ,  H04L65/605

Abstract: Devices, softwares and methods generate, in real time, indexing metadata for select portions of a telephone conversation or conference. The indexing metadata is generated responsive to inputs received while the conversation is being recorded live. The inputs are either by a user pressing a soft key on a telephone device, or by a voice conference bridge determining who is the dominant speaker in a multi-party conference.

Abstract translation: 设备，软件和方法实时地为电话对话或会议的选择部分生成索引元数据。 索引元数据是响应于在现场记录对话期间接收到的输入而生成的。 输入是通过用户按下电话设备上的软键，或通过语音会议桥来确定谁是多方会议中的主要扬声器。

公开(公告)号：US07180997B2

公开(公告)日：2007-02-20

申请号：US10236484

申请日：2002-09-06

Applicant: Michael E. Knappe

Inventor： Michael E. Knappe

IPC: H04M3/56

CPC classification number: H04M3/42187 ,  H04M3/2227 ,  H04M3/2236 ,  H04M3/561 ,  H04M3/568 ,  H04M3/569 ,  H04M7/006 ,  H04M7/1205 ,  H04M2207/203

Abstract: A system and method for improving the intelligibility of a moderator during a multi-party communication session includes receiving a plurality of participant voice streams from a plurality of respective conference participants. An incoming moderator voice stream may be received from a moderator. The plurality of participant voice streams and the moderator voice stream are transmitted such that the intelligibility of the moderator voice stream is improved relative to at least one of the participant voice streams.

Abstract translation: 一种用于在多方通信会话期间提高主持人的可懂度的系统和方法包括从多个相应的会议参与者接收多个参与者语音流。 可以从主持人接收传入的主持人语音流。 发送多个参与者语音流和主持人语音流，使得相对于参与者语音流中的至少一个提高了主持人语音流的可懂度。

公开(公告)号：US06973184B1

公开(公告)日：2005-12-06

申请号：US09614535

申请日：2000-07-11

Applicant: Shmuel Shaffer ,  Michael E. Knappe

Inventor： Shmuel Shaffer ,  Michael E. Knappe

IPC: H04M9/08 ,  H04R27/00 ,  H04S5/00

CPC classification number: H04R27/00 ,  G10L19/008

Abstract: Systems and methods are disclosed for packet voice conferencing. An encoding system accepts two sound field signals, representing the same sound field sampled at two spatially-separated points. The relative delay between the two sound field signals is detected over a given time interval. The sound field signals are combined and then encoded as a single audio signal, e.g., by a method suitable for monophonic VoIP. The encoded audio payload and the relative delay are placed in one or more packets and sent to a decoding device via the packet network. The decoding device uses the relative delay to drive a playout splitter—once the encoded audio payload has been decoded, the playout splitter creates multiple presentation channels by inserting the transmitted relative delay in the decoded signal for one (or more) of the presentation channels. The listener thus perceives a speaker's voice as originating from a location related to the speaker's physical position at the other end of the conference. An advantage of these embodiments is that a pseudo-stereo conference can be conducted with virtually the same bandwidth as a monophonic conference.

Abstract translation: 公开了用于分组语音会议的系统和方法。 编码系统接收两个声场信号，表示在两个空间分离点采样的相同声场。 在给定的时间间隔内检测两个声场信号之间的相对延迟。 声场信号被组合，然后被编码为单个音频信号，例如通过适合于单声道VoIP的方法。 编码的音频有效载荷和相对延迟被放置在一个或多个分组中，并且经由分组网络被发送到解码装置。 解码装置使用相对延迟来驱动播出分离器 - 一旦编码的音频有效载荷已被解码，播放分离器通过将一个（或更多个）演示频道的解码信号中插入传输的相对延迟来创建多个显示频道。 因此，听众将演讲者的声音从与会议另一端的演讲人的身体位置相关的位置发出。 这些实施例的优点是可以以与单声道会议几乎相同的带宽进行伪立体会议。

公开(公告)号：US20120120797A1

公开(公告)日：2012-05-17

申请号：US13355945

申请日：2012-01-23

Applicant: Paul S. Hahn ,  Michael E. Knappe ,  Richard A. Dunlap ,  Luke K. Surazski

Inventor： Paul S. Hahn ,  Michael E. Knappe ,  Richard A. Dunlap ,  Luke K. Surazski

IPC: H04L12/26

CPC classification number: H04J3/0632 ,  H04Q2213/13034 ,  H04Q2213/13389

Abstract: According to one embodiment of the invention, a method for managing time-sensitive packetized data streams at a receiver includes receiving a time-sensitive packet of a data stream, analyzing an energy level of a payload signal of the packet, and determining whether to drop the packet based on the energy level of the payload signal.

Abstract translation: 根据本发明的一个实施例，一种用于在接收器处管理时间敏感分组化数据流的方法包括：接收数据流的时间敏感分组，分析分组的有效载荷信号的能级，并确定是否丢弃 该分组基于有效载荷信号的能级。

公开(公告)号：US07058168B1

公开(公告)日：2006-06-06

申请号：US09751799

申请日：2000-12-29

Applicant: Michael E. Knappe ,  Shmuel Shaffer

Inventor： Michael E. Knappe ,  Shmuel Shaffer

IPC: H04M3/42 ,  H04L12/16 ,  H04Q11/00 ,  G06F15/16

CPC classification number: H04L29/06027 ,  H04L12/1822 ,  H04L65/4038 ,  H04M3/56 ,  H04M3/564 ,  H04Q3/0025

Abstract: A method and system for participant control of privacy during a multiparty communication session includes receiving a request from a first participant to a multiparty communication connection for a sidebar between the first participant and a second participant to the multiparty communication connection. The sidebar is provided by at least substantially eliminating voice streams generated by the first participant and the second participant from conference output streams generated for a set of remaining participants to the multiparty communication connection.

Abstract translation: 一种用于在多方通信会话期间参与者控制隐私的方法和系统包括从第一参与者接收针对第一参与者和第二参与者之间的侧边栏的多方通信连接到多方通信连接的请求。 侧栏通过至少基本上消除由第一参与者和第二参与者产生的语音流从为一组剩余参与者生成的会议输出流提供到多方通信连接。

公开(公告)号：US06999560B1

公开(公告)日：2006-02-14

申请号：US09340987

申请日：1999-06-28

Applicant: Kevin J. Connor ,  Michael E. Knappe ,  Ramanathan T. Jagadeesan

Inventor： Kevin J. Connor ,  Michael E. Knappe ,  Ramanathan T. Jagadeesan

IPC: H04M1/24 ,  H04M3/08 ,  H04M3/22

CPC classification number: H04M3/002 ,  H04M3/22 ,  H04M3/323

Abstract: A test system measures performance of telephone network echo cancellers using a primary criterion of estimated user annoyance due to audible returned echo. The invention generates live telephone calls, uses real speech samples as stimulus signals and provides tail-circuit emulation using actual measured telephone tail-circuit impulse responses. These features provide better ‘real-life’ test conditions for the echo canceller system under test than current ITU standard test methods. Two methods are employed for echo canceller performance evaluation via metrics of estimated user annoyance due to echo. Energy-based method employs point-by-point comparison of talker speech and talker echo signal energy envelopes and uses variable energy thresholds for estimation of echo audibility. A perceptual-model based method uses a Perceptual Speech Distortion Metric (PSDM), such as ITU P.861, in an unique configuration to estimate user annoyance due to audible echo. Echo canceller performance is tested under both single-talk and double-talk conditions. Innovative application of the PSDM method in double-talk tests allow estimation of quality of received double-talk speech.

Abstract translation: 测试系统使用由于可听见的返回回声估计的用户烦恼的主要标准来测量电话网络回声消除器的性能。 本发明产生实时电话呼叫，使用实际语音样本作为刺激信号，并使用实际测量的电话尾循环脉冲响应提供尾巴仿真。 这些功能为被测回声消除器系统提供比当前国际电联标准测试方法更好的“现实生活”测试条件。 回波消除器性能评估采用两种方法，通过由于回声引起的估计用户烦恼的度量。 基于能量的方法使用讲话者语音和讲话者回波信号能量包络的逐点比较，并且使用可变能量阈值来估计回声可听度。 基于感知模型的方法使用感知语音失真度量（PSDM），例如ITU P.861，以独特的配置来估计由于可听见的回声引起的用户烦恼。 回声消除器性能在单声道和双通道条件下进行测试。 PSDM方法在双向通话测试中的创新应用允许估计接收的双方通话语音的质量。