专利快速检索-快速检索全球专利，免费商用专利数据库-IPRDB

1. 发明授权

US07715404B2 Method and apparatus for controlling a voice over internet protocol (VoIP) decoder with an adaptive jitter buffer 有权
标题翻译：用于利用自适应抖动缓冲器来控制具有互联网协议语音（VoIP）解码器的方法和装置
公开(公告)号：US07715404B2
公开(公告)日：2010-05-11
申请号：US11940505
申请日：2007-11-15
申请人： Richard L. Zinser, Jr. , Martin W. Egan
发明人： Richard L. Zinser, Jr. , Martin W. Egan
IPC分类号： H04L12/28 , H04L12/56 , H04J3/24
CPC分类号： H04L65/607 , H04L65/608
摘要： A method and apparatus that controls a Voice over Internet Protocol (VoIP) decoder in a communication device is disclosed. The method may include determining if a packet has been received, and if a packet has been received, determining if a receive error has occurred, and if a receive error has not occurred, setting a counter that counts a number of sequential bad frames to a value of zero, decoding the received packet, and sending the decoded packet to an audio queue for presentation to a user of the communication device.
摘要翻译：公开了一种在通信设备中控制因特网协议语音（VoIP）解码器的方法和装置。该方法可以包括确定是否已经接收到分组，并且如果已经接收到分组，则确定是否已经发生接收错误，并且如果没有发生接收错误，则将计数连续不良帧的数量的计数器设置为值为零，对接收到的分组进行解码，以及将解码的分组发送到音频队列以呈现给通信设备的用户。

2. 发明授权

US07489635B2 Routing cost based network congestion control for quality of service 有权
标题翻译：基于路由成本的网络拥塞控制服务质量
公开(公告)号：US07489635B2
公开(公告)日：2009-02-10
申请号：US10950149
申请日：2004-09-24
申请人： Scott C. Evans , Marc R. Pearlman , Michael J. Hartman , Asavari Rothe , Manuel A. Leiva , Martin W. Egan
发明人： Scott C. Evans , Marc R. Pearlman , Michael J. Hartman , Asavari Rothe , Manuel A. Leiva , Martin W. Egan
IPC分类号： H04J1/16
CPC分类号： H04L41/5025 , H04L41/5003 , H04L45/00 , H04L45/02 , H04L45/121 , H04L45/302 , H04L47/10 , H04L47/11 , H04L47/12 , H04L47/14 , H04L47/2408 , H04L47/2416 , H04L47/283 , H04L47/32 , H04W28/08 , H04W28/14 , H04W40/00 , H04W80/00 , H04W84/18
摘要： A method and system of congestion control in a network are provided. A required quality of service (QoS) parameter, such as a maximum allowable latency, for a packet received at a queue in the network, and a route cost metric, such as accumulated and estimated latency, are determined, and the packet is either discarded if the route cost metric exceeds the required QoS parameter, or a discard bias value is set for the packet. Also, if the required QoS parameter exceeds the route cost metric, the method includes determining whether a congestion condition exists in the network, and if the congestion condition exists, biasing the packet for discard based on its latency if the route cost metric for the packet exceeds a threshold. The network may be an IP network, and a network such as a mobile ad hoc network (MANET).
摘要翻译：提供网络中拥塞控制的方法和系统。确定在网络中的队列处接收到的分组的所需服务质量（QoS）参数（例如，最大允许延迟）以及路由成本度量（例如累积和估计的等待时间），并且该分组被丢弃如果路由成本度量超过所需的QoS参数，或者为分组设置丢弃偏移值。此外，如果所需QoS参数超过路由成本度量，则该方法包括确定网络中是否存在拥塞状况，并且如果存在拥塞状况，则如果分组的路由开销度量，则基于其延迟来偏移丢弃的分组超过阈值。网络可以是IP网络，以及诸如移动自组织网络（MANET）的网络。

3. 发明申请

US20110299399A1 METHOD AND APPARATUS FOR CONGESTION CONTROL 有权
标题翻译：控制方法和装置
公开(公告)号：US20110299399A1
公开(公告)日：2011-12-08
申请号：US12795172
申请日：2010-06-07
申请人： Martin W. EGAN , Richard Louis Zinser , Bruce Gordon Barnett
发明人： Martin W. EGAN , Richard Louis Zinser , Bruce Gordon Barnett
IPC分类号： H04L12/26
CPC分类号： H04L47/2416 , H04L45/28 , H04L47/10 , H04L47/193 , H04L47/263 , H04L47/283 , H04L47/38 , Y02D50/10
摘要： Aspects of the disclosure provide a method of congestion control. The method includes transmitting by a first apparatus to a second apparatus a packet stream corresponding to stream data generated by a variable data rate application, determining a round-trip-time of a packet in the packet stream, updating a congestion measure based on the round-trip-time, and informing the variable data rate application based on the congestion measure to adjust a data rate for generating the stream data.
摘要翻译：本公开的方面提供了一种拥塞控制的方法。所述方法包括由第一设备向第二设备发送与由可变数据速率应用产生的流数据相对应的分组流，确定分组流中分组的往返时间，基于所述轮回更新拥塞度量并且基于拥塞度量通知可变数据速率应用以调整用于生成流数据的数据速率。

4. 发明授权

US07970603B2 Method and apparatus for managing speech decoders in a communication device 有权
标题翻译：用于在通信设备中管理语音解码器的方法和装置
公开(公告)号：US07970603B2
公开(公告)日：2011-06-28
申请号：US11940435
申请日：2007-11-15
申请人： Richard L. Zinser, Jr. , Martin W. Egan
发明人： Richard L. Zinser, Jr. , Martin W. Egan
IPC分类号： G10L19/00
CPC分类号： G10L19/16 , G10L19/24
摘要： A method and apparatus that manages speech decoders in a communication device may include detecting a change in transmission rate from a higher rate to a lower rate, decoding and shifting a first, second and third received first decoder set of frame parameters, generating a first decoder output audio frame from the previously shifted frame parameters, generating a first, second and third second decoder audio fill frame, the second decoder being a higher rate decoder than first decoder, outputting a first and second second decoder audio fill frame, combining the first decoder audio frame and the third second decoder audio fill frame with overlapping triangular windows, and outputting combined first decoder and second decoder frames to an audio buffer for subsequent transmission to a user of the communication device. In an alternative embodiment, another method may include detecting and processing a change in transmission rate from a lower rate to a higher rate.
摘要翻译：管理通信设备中的语音解码器的方法和装置可以包括检测传输速率从较高速率到较低速率的变化，解码和移位第一，第二和第三接收到的第一解码器帧参数组，生成第一解码器从先前移位的帧参数输出音频帧，产生第一，第二和第三解码器音频填充帧，第二解码器是比第一解码器更高的速率解码器，输出第一和第二第二解码器音频填充帧，组合第一解码器音频帧和具有重叠三角形窗口的第三第二解码器音频填充帧，并且将组合的第一解码器和第二解码器帧输出到音频缓冲器，以便随后向通信设备的用户发送。在替代实施例中，另一种方法可以包括检测和处理传输速率从较低速率到较高速率的变化。

5. 发明授权

US08340126B2 Method and apparatus for congestion control 有权
标题翻译：用于拥塞控制的方法和装置
公开(公告)号：US08340126B2
公开(公告)日：2012-12-25
申请号：US12795172
申请日：2010-06-07
申请人： Martin W. Egan , Richard Louis Zinser , Bruce Gordon Barnett
发明人： Martin W. Egan , Richard Louis Zinser , Bruce Gordon Barnett
IPC分类号： H04J3/16
CPC分类号： H04L47/2416 , H04L45/28 , H04L47/10 , H04L47/193 , H04L47/263 , H04L47/283 , H04L47/38 , Y02D50/10
摘要： Aspects of the disclosure provide a method of congestion control. The method includes transmitting by a first apparatus to a second apparatus a packet stream corresponding to stream data generated by a variable data rate application, determining a round-trip-time of a packet in the packet stream, updating a congestion measure based on the round-trip-time, and informing the variable data rate application based on the congestion measure to adjust a data rate for generating the stream data.
摘要翻译：本公开的方面提供了一种拥塞控制的方法。所述方法包括由第一设备向第二设备发送与由可变数据速率应用产生的流数据相对应的分组流，确定分组流中分组的往返时间，基于所述轮回更新拥塞度量并且基于拥塞度量通知可变数据速率应用以调整用于生成流数据的数据速率。

6. 发明授权

US07738361B2 Method and apparatus for generating fill frames for voice over internet protocol (VoIP) applications 有权
标题翻译：用于产生用于网际语音协议（VoIP）应用的填充帧的方法和装置
公开(公告)号：US07738361B2
公开(公告)日：2010-06-15
申请号：US11940591
申请日：2007-11-15
申请人： Richard L. Zinser, Jr. , Martin W. Egan
发明人： Richard L. Zinser, Jr. , Martin W. Egan
IPC分类号： G06F11/00
CPC分类号： H04L65/607 , G10L19/005 , H04L65/1059 , H04L65/80
摘要： A method and apparatus that generates fill frames for Voice over Internet Protocol (VoIP) applications in a communication device is disclosed. The method may include determining if there is a lost frame in a received communication, wherein if it is determined that there is a lost frame, setting a frame loss flag and storing the frame loss flag in the frame loss history register, shifting a loss history register, a line spectral frequency (LSF) history register, a voicing cutoff (VCUT) history register, a pitch history register, and a root mean squared (RMS) gain history register, wherein the loss history register, the LSF history register, the VCUT history register, the pitch history register, and the RMS history register include at least three registers, the three registers being a newest, a middle and an oldest registers, reading the frame loss flag into a newest loss history register, determining contents of the middle register of each of the LSF history register, the VCUT history register, the pitch history register, and the RMS history register, and sending the contents of the middle registers to a synthesizer to generate an output speech signal.
摘要翻译：公开了一种在通信设备中产生用于因特网协议语音（VoIP）应用的填充帧的方法和装置。该方法可以包括确定接收到的通信中是否存在丢失帧，其中如果确定存在丢失帧，则设置帧丢失标志并将帧丢失标志存储在帧丢失历史寄存器中，将丢失历史寄存器，线谱频率（LSF）历史寄存器，语音切换（VCUT）历史寄存器，音调历史寄存器和均方根（RMS）增益历史寄存器，其中丢失历史寄存器，LSF历史寄存器， VCUT历史寄存器，音调历史寄存器和RMS历史寄存器包括至少三个寄存器，三个寄存器是最新的，中间的和最旧的寄存器，将帧丢失标志读入最新的丢失历史寄存器，确定每个LSF历史寄存器的中间寄存器，VCUT历史寄存器，音调历史寄存器和RMS历史寄存器，并将中间寄存器的内容发送到合成器，以生成输出语音信号。

7. 发明申请

US20090132246A1 METHOD AND APPARATUS FOR GENERATING FILL FRAMES FOR VOICE OVER INTERNET PROTOCOL (VoIP) APPLICATIONS 有权
标题翻译：用于通过互联网协议语音（VoIP）应用产生语音框架的方法和装置
公开(公告)号：US20090132246A1
公开(公告)日：2009-05-21
申请号：US11940591
申请日：2007-11-15
申请人： Richard L. ZINSER, JR. , Martin W. Egan
发明人： Richard L. ZINSER, JR. , Martin W. Egan
IPC分类号： G10L19/00
CPC分类号： H04L65/607 , G10L19/005 , H04L65/1059 , H04L65/80
摘要： A method and apparatus that generates fill frames for Voice over Internet Protocol (VoIP) applications in a communication device is disclosed. The method may include determining if there is a lost frame in a received communication, wherein if it is determined that there is a lost frame, setting a frame loss flag and storing the frame loss flag in the frame loss history register, shifting a loss history register, a line spectral frequency (LSF) history register, a voicing cutoff (VCUT) history register, a pitch history register, and a root mean squared (RMS) gain history register, wherein the loss history register, the LSF history register, the VCUT history register, the pitch history register, and the RMS history register include at least three registers, the three registers being a newest, a middle and an oldest registers, reading the frame loss flag into a newest loss history register, determining contents of the middle register of each of the LSF history register, the VCUT history register, the pitch history register, and the RMS history register, and sending the contents of the middle registers to a synthesizer to generate an output speech signal.
摘要翻译：公开了一种在通信设备中产生用于因特网协议语音（VoIP）应用的填充帧的方法和装置。该方法可以包括确定接收到的通信中是否存在丢失帧，其中如果确定存在丢失帧，则设置帧丢失标志并将帧丢失标志存储在帧丢失历史寄存器中，将丢失历史寄存器，线谱频率（LSF）历史寄存器，语音切换（VCUT）历史寄存器，音调历史寄存器和均方根（RMS）增益历史寄存器，其中丢失历史寄存器，LSF历史寄存器， VCUT历史寄存器，音调历史寄存器和RMS历史寄存器包括至少三个寄存器，三个寄存器是最新的，中间的和最旧的寄存器，将帧丢失标志读入最新的丢失历史寄存器，确定每个LSF历史寄存器的中间寄存器，VCUT历史寄存器，音调历史寄存器和RMS历史寄存器，并将中间寄存器的内容发送到合成器，以生成输出语音信号。

8. 发明申请

US20090132244A1 METHOD AND APPARATUS FOR CONTROLLING A VOICE OVER INTERNET PROTOCOL (VoIP) DECODER WITH AN ADAPTIVE JITTER BUFFER 有权
标题翻译：用于通过自适应抖动缓冲器控制互联网协议（VoIP）解码器的语音的方法和装置
公开(公告)号：US20090132244A1
公开(公告)日：2009-05-21
申请号：US11940505
申请日：2007-11-15
申请人： Richard L. ZINSER, JR. , Martin W. EGAN
发明人： Richard L. ZINSER, JR. , Martin W. EGAN
IPC分类号： G10L21/02
CPC分类号： H04L65/607 , H04L65/608
摘要： A method and apparatus that controls a Voice over Internet Protocol (VoIP) decoder in a communication device is disclosed. The method may include determining if a packet has been received, and if a packet has been received, determining if a receive error has occurred, and if a receive error has not occurred, setting a counter that counts a number of sequential bad frames to a value of zero, decoding the received packet, and sending the decoded packet to an audio queue for presentation to a user of the communication device.
摘要翻译：公开了一种在通信设备中控制因特网协议语音（VoIP）解码器的方法和装置。该方法可以包括确定是否已经接收到分组，并且如果已经接收到分组，则确定是否已经发生接收错误，并且如果没有发生接收错误，则将计数连续不良帧的数量的计数器设置为值为零，对接收到的分组进行解码，以及将解码的分组发送到音频队列以呈现给通信设备的用户。

9. 发明申请

US20090132240A1 METHOD AND APPARATUS FOR MANAGING SPEECH DECODERS 有权
标题翻译：管理语音解码器的方法和设备
公开(公告)号：US20090132240A1
公开(公告)日：2009-05-21
申请号：US11940435
申请日：2007-11-15
申请人： Richard L. Zinser, JR. , Martin W. Egan
发明人： Richard L. Zinser, JR. , Martin W. Egan
IPC分类号： G10L19/00
CPC分类号： G10L19/16 , G10L19/24
摘要： A method and apparatus that manages speech decoders in a communication device is disclosed. The method may include detecting a change in transmission rate from a higher rate to a lower rate, decoding and shifting a first, second and third received first decoder set of frame parameters, generating a first decoder output audio frame from the previously shifted frame parameters, generating a first, second and third second decoder audio fill frame, the second decoder being a higher rate decoder than first decoder, outputting a first and second second decoder audio fill frame, combining the first decoder audio frame and the third second decoder audio fill frame with overlapping triangular windows, and outputting combined first decoder and second decoder frames to an audio buffer for subsequent transmission to a user of the communication device. In an alternative embodiment, another method may include detecting and processing a change in transmission rate from a lower rate to a higher rate.
摘要翻译：公开了一种在通信设备中管理语音解码器的方法和装置。该方法可以包括检测从较高速率到较低速率的传输速率的变化，解码和移位第一，第二和第三接收到的第一解码器帧参数组，从先前移位的帧参数生成第一解码器输出音频帧，产生第一，第二和第三解码器音频填充帧，第二解码器是比第一解码器更高的速率解码器，输出第一和第二解码器音频填充帧，组合第一解码器音频帧和第三第二解码器音频填充帧具有重叠的三角形窗口，并且将组合的第一解码器和第二解码器帧输出到音频缓冲器，以便随后向通信设备的用户发送。在替代实施例中，另一种方法可以包括检测和处理传输速率从较低速率到较高速率的变化。

你已经成功收藏专利！

检索式保存成功!

IPRDB

热门服务

关于我们

友情链接

联系方式