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1. 发明授权

US08811602B2 Full duplex speakerphone design using acoustically compensated speaker distortion 有权
标题翻译：使用声学补偿扬声器失真的全双工免提电话设计
公开(公告)号：US08811602B2
公开(公告)日：2014-08-19
申请号：US13174148
申请日：2011-06-30
申请人： Prakash Khanduri , Nelson Sollenberger , Huaiyu Zeng
发明人： Prakash Khanduri , Nelson Sollenberger , Huaiyu Zeng
IPC分类号： H04M9/08 , H04B3/20
CPC分类号： H04M9/082
摘要： A telecommunication system including a fall duplex speakerphone, comprising a first microphone to generate a coupled signal including uplink information and non-linear distortion sensed by the first microphone in a speaker phone mode, a second microphone to generate a reference signal including downlink information and the non-linear distortion sensed by the second microphone in the speaker phone mode, and an acoustic echo canceller (AEC) to receive the coupled signal from the first microphone, to receive the reference signal from the second microphone, and to cancel out the non-linear distortion included in the coupled signal based on the non-linear distortion included in the reference signal.
摘要翻译：一种包括秋季双工免提电话的电信系统，包括：第一麦克风，用于产生包括在扬声器电话模式中由所述第一麦克风感测的上行链路信息和非线性失真的耦合信号;第二麦克风，用于生成包括下行链路信息的参考信号;以及在扬声器电话模式下由第二麦克风感测到的非线性失真，以及声学回声消除器（AEC），用于接收来自第一麦克风的耦合信号，以接收来自第二麦克风的参考信号，基于参考信号中包含的非线性失真，包含在耦合信号中的线性失真。

2. 发明授权

US08687797B2 Method and system for a dual echo canceller 有权
标题翻译：双回波消除器的方法和系统
公开(公告)号：US08687797B2
公开(公告)日：2014-04-01
申请号：US12474065
申请日：2009-05-28
申请人： Hanks Zeng , Prakash Khanduri , Ken Kasiske , Ronish Patel , Nelson Sollenberger
发明人： Hanks Zeng , Prakash Khanduri , Ken Kasiske , Ronish Patel , Nelson Sollenberger
IPC分类号： H04M9/08
CPC分类号： H04M9/082
摘要： Methods and systems for a dual echo canceller (EC) are disclosed and may include cancelling echo in utilizing a dual echo canceller, wherein said dual echo canceller includes an active echo canceller and an adaptive echo canceller. Filter coefficients may be copied from the adaptive echo canceller to the active echo canceller for the cancellation, based on whether said adaptive echo canceller has converged. The coefficients may be copied utilizing copy logic, which may comprise divergence detection and/or echo path change detection. The coefficients may be reset to default settings utilizing the copy logic. The coefficients may be calculated utilizing normalized block least mean squares (NBLMS), and may be calculated when the NBLMS is enabled by update logic. The coefficients may be calculated utilizing linear predictive coefficient (LPC) filtered uplink and downlink signals.
摘要翻译：公开了用于双回波消除器（EC）的方法和系统，并且可以包括利用双回波消除器来消除回波，其中所述双回波消除器包括主动回波消除器和自适应回波消除器。基于所述自适应回波消除器是否收敛，可以将滤波器系数从自适应回波消除器复制到用于消除的有源回波消除器。可以利用可能包括发散检测和/或回波路径变化检测的复制逻辑来复制系数。使用复制逻辑可以将系数重置为默认设置。可以使用归一化块最小均方（NBLMS）来计算系数，并且可以在更新逻辑启用NBLMS时计算系数。可以使用线性预测系数（LPC）滤波的上行链路和下行链路信号来计算系数。

3. 发明申请

US20100304679A1 Method and System For Echo Estimation and Cancellation 审中-公开
标题翻译：回波估计和取消方法与系统
公开(公告)号：US20100304679A1
公开(公告)日：2010-12-02
申请号：US12474061
申请日：2009-05-28
申请人： Hanks Zeng , Prakash Khanduri , Ken Kasiske , Ronish Patel , Nelson Sollenberger
发明人： Hanks Zeng , Prakash Khanduri , Ken Kasiske , Ronish Patel , Nelson Sollenberger
IPC分类号： H04B15/00
CPC分类号： H04M9/082 , H04B7/015 , H04M1/6066
摘要： Methods and systems for echo estimation and cancellation are disclosed and may include estimating combined echo return loss and echo return loss enhancement (ERL+ERLE). A subband gain vector may be calculated utilizing non-linear processing, subband analysis, and the estimated ERL+ERLE to mitigate residual echo. ERL+ERLE may be estimated by averaging a difference in DL and UL signals. A maximum value of ERL+ERLE may be determined over a period of time. A non-linear distortion adjustment factor may be estimated for ERL+ERLE. An ERL+ERLE estimation error may be calibrated specific to the wireless device. The estimating of ERL+ERLE may be suspended briefly after a transition in the DL or UL signals. Comfort noise may be added to mask the residual echo, which may be mitigated following a dual echo canceller. The estimating may be suspended when the DL signal is not present. A noise level may be included in the gain calculation.
摘要翻译：公开了用于回波估计和消除的方法和系统，并且可以包括估计组合的回波回波损耗和回波回波损耗增强（ERL + ERLE）。可以使用非线性处理，子带分析和估计的ERL + ERLE来计算子带增益矢量以减轻残余回波。可以通过对DL和UL信号中的差进行平均来估计ERL + ERLE。 ERL + ERLE的最大值可以在一段时间内确定。可以对ERL + ERLE估计非线性失真调整因子。 ERL + ERLE估计误差可能会针对无线设备进行校准。 ERL + ERLE的估计可以在DL或UL信号转换后暂时暂停。可以添加舒适噪声来掩蔽残留回波，这可以在双回波消除器之后被缓解。当DL信号不存在时，可以暂停估计。噪声电平可能包含在增益计算中。

4. 发明授权

US08626516B2 Method and system for dynamic range control in an audio processing system 有权
标题翻译：音频处理系统中动态范围控制的方法和系统
公开(公告)号：US08626516B2
公开(公告)日：2014-01-07
申请号：US12367854
申请日：2009-02-09
申请人： Hanks Zeng , Prakash Khanduri , Ken Kasiske , Ronish Patel , Nelson Sollenberger
发明人： Hanks Zeng , Prakash Khanduri , Ken Kasiske , Ronish Patel , Nelson Sollenberger
IPC分类号： G10L19/00 , G10L21/00
CPC分类号： H04R3/00 , H03G9/025 , H04R2430/03
摘要： Methods and systems for dynamic range control in an audio processing system are disclosed and may include controlling a dynamic range of an audio signal by expanding the dynamic range utilizing a dynamic expander, and dividing the audio signal into a plurality of frequency bands. Each of the bands may be individually compressed utilizing a multi-band compressor. A sum of the individually compressed frequency bands may be compressed utilizing a full-band compressor. The audio signal may be filtered utilizing a pre-emphasis filter, such as an infinite impulse response filter and may be divided into frequency bands utilizing one or more finite impulse response filters and/or delay modules. The dynamic expander may include adaptive thresholds and an envelope detector. Each of the frequency bands may be compressed utilizing syllabic compression in the multi-band compressor. The compressed sum of compressed plurality of bands may be processed utilizing an audio CODEC.
摘要翻译：公开了用于音频处理系统中的动态范围控制的方法和系统，并且可以包括通过使用动态扩展器扩展动态范围并且将音频信号划分成多个频带来控制音频信号的动态范围。可以使用多频带压缩机来单独压缩每个频带。可以使用全频带压缩器压缩单独压缩的频带的总和。可以使用诸如无限脉冲响应滤波器的预加重滤波器来对音频信号进行滤波，并且可以使用一个或多个有限脉冲响应滤波器和/或延迟模块将音频信号分成频带。动态扩展器可以包括自适应阈值和包络检测器。可以在多频带压缩器中使用音节压缩来压缩每个频带。可以使用音频CODEC来处理压缩的多个频带的压缩和。

5. 发明申请

US20100303228A1 METHOD AND SYSTEM FOR A DUAL ECHO CANCELLER 有权
标题翻译：一种双重收音机的方法与系统
公开(公告)号：US20100303228A1
公开(公告)日：2010-12-02
申请号：US12474065
申请日：2009-05-28
申请人： Hanks Zeng , Prakash Khanduri , Ken Kasiske , Ronish Patel , Nelson Sollenberger
发明人： Hanks Zeng , Prakash Khanduri , Ken Kasiske , Ronish Patel , Nelson Sollenberger
IPC分类号： H04M9/08
CPC分类号： H04M9/082
摘要： Methods and systems for a dual echo canceller (EC) are disclosed and may include cancelling echo in utilizing a dual echo canceller, wherein said dual echo canceller includes an active echo canceller and an adaptive echo canceller. Filter coefficients may be copied from the adaptive echo canceller to the active echo canceller for the cancellation, based on whether said adaptive echo canceller has converged. The coefficients may be copied utilizing copy logic, which may comprise divergence detection and/or echo path change detection. The coefficients may be reset to default settings utilizing the copy logic. The coefficients may be calculated utilizing normalized block least mean squares (NBLMS), and may be calculated when the NBLMS is enabled by update logic. The coefficients may be calculated utilizing linear predictive coefficient (LPC) filtered uplink and downlink signals.
摘要翻译：公开了用于双回波消除器（EC）的方法和系统，并且可以包括利用双回波消除器来消除回波，其中所述双回波消除器包括主动回波消除器和自适应回波消除器。基于所述自适应回波消除器是否收敛，可以将滤波器系数从自适应回波消除器复制到用于消除的有效回波消除器。可以利用可能包括发散检测和/或回波路径变化检测的复制逻辑来复制系数。使用复制逻辑可以将系数重置为默认设置。可以使用归一化块最小均方（NBLMS）来计算系数，并且可以在更新逻辑启用NBLMS时计算系数。可以使用线性预测系数（LPC）滤波的上行链路和下行链路信号来计算系数。

6. 发明申请

US20100204996A1 METHOD AND SYSTEM FOR DYNAMIC RANGE CONTROL IN AN AUDIO PROCESSING SYSTEM 有权
标题翻译：音频处理系统中动态范围控制的方法和系统
公开(公告)号：US20100204996A1
公开(公告)日：2010-08-12
申请号：US12367854
申请日：2009-02-09
申请人： Hanks Zeng , Prakash Khanduri , Ken Kasiske , Ronish Patel , Nelson Sollenberger
发明人： Hanks Zeng , Prakash Khanduri , Ken Kasiske , Ronish Patel , Nelson Sollenberger
IPC分类号： G10L21/00
CPC分类号： H04R3/00 , H03G9/025 , H04R2430/03
摘要： Methods and systems for dynamic range control in an audio processing system are disclosed and may include controlling a dynamic range of an audio signal by expanding the dynamic range utilizing a dynamic expander, and dividing the audio signal into a plurality of frequency bands. Each of the bands may be individually compressed utilizing a multi-band compressor. A sum of the individually compressed frequency bands may be compressed utilizing a full-band compressor. The audio signal may be filtered utilizing a pre-emphasis filter, such as an infinite impulse response filter and may be divided into frequency bands utilizing one or more finite impulse response filters and/or delay modules. The dynamic expander may include adaptive thresholds and an envelope detector. Each of the frequency bands may be compressed utilizing syllabic compression in the multi-band compressor. The compressed sum of compressed plurality of bands may be processed utilizing an audio CODEC.
摘要翻译：公开了用于音频处理系统中的动态范围控制的方法和系统，并且可以包括通过使用动态扩展器扩展动态范围并且将音频信号划分成多个频带来控制音频信号的动态范围。可以使用多频带压缩机来单独压缩每个频带。可以使用全频带压缩器压缩单独压缩的频带的总和。可以使用诸如无限脉冲响应滤波器的预加重滤波器来对音频信号进行滤波，并且可以使用一个或多个有限脉冲响应滤波器和/或延迟模块将音频信号分成频带。动态扩展器可以包括自适应阈值和包络检测器。可以在多频带压缩器中使用音节压缩来压缩每个频带。可以使用音频CODEC来处理压缩的多个频带的压缩和。

7. 发明授权

US08767687B2 Method and system for endpoint based architecture for VoIP access points 有权
标题翻译：用于VoIP接入点的基于端点的架构的方法和系统
公开(公告)号：US08767687B2
公开(公告)日：2014-07-01
申请号：US12433937
申请日：2009-05-01
申请人： Prakash Khanduri
发明人： Prakash Khanduri
IPC分类号： H04W4/00
CPC分类号： H04L65/103 , H04M7/0066 , H04W88/06 , H04W88/16 , H04W88/181 , H04W92/02
摘要： A VoIP access point may be operable to provide VoIP servicing to a plurality of wireless audio endpoint devices. The VoIP access point may extract VoIP audio data received via IP backbone and communicating the extracted audio data as non-VoIP formatted data to the wireless audio endpoint devices. In the uplink direction, the VoIP access point may receive non-VoIP formatted audio data from the wireless audio endpoint devices and pack the received data into IP packets for VoIP communication. The VoIP access point may also be operable to perform PCM encoding/decoding operations during VoIP servicing operations. The wireless audio endpoint devices may perform audio processing during VoIP communications via the VoIP access point. One or more intermediary communication devices may be utilized to route messaging between the VoIP access point and at least some of the wireless audio endpoint devices.
摘要翻译： VoIP接入点可以用于向多个无线音频端点设备提供VoIP服务。 VoIP接入点可以提取通过IP骨干接收的VoIP音频数据，并将所提取的音频数据作为非VoIP格式的数据传送到无线音频端点设备。在上行方向，VoIP接入点可以从无线音频端点设备接收非VoIP格式的音频数据，并将接收到的数据打包成用于VoIP通信的IP分组。 VoIP接入点也可以用于在VoIP服务操作期间执行PCM编码/解码操作。无线音频端点设备可以经由VoIP接入点进行VoIP通信期间的音频处理。一个或多个中间通信设备可以用于在VoIP接入点和至少一些无线音频端点设备之间路由消息传递。

8. 发明授权

US09154596B2 Method and system for audio system volume control 有权
标题翻译：音频系统音量控制的方法和系统
公开(公告)号：US09154596B2
公开(公告)日：2015-10-06
申请号：US12548758
申请日：2009-08-27
申请人： Prakash Khanduri
发明人： Prakash Khanduri
IPC分类号： H04M1/00 , H04M1/60
CPC分类号： H04M1/6016
摘要： A communication device may determine total gain for a downlink processing path via the communication device. The total gain may be determined based on determination of audio requirements for each of a plurality of audio processing devices supported via the communication device, during the downlink processing path. The communication device is operable to determine a default mix signal gain, a downlink gain adjustment and a calibration gain based on the determined audio requirements, and to calculate the total gain for the downlink processing path based on the determined default mix signal gain, the downlink gain adjustment and the calibration gain. The determination of the total gain may be based on determination of a telephony state in the communication device. The calibration gain may be determined during calibration of the communication device, whereas the downlink gain adjustment may be calculated based on an adjustment step-size and/or a mode based gain index.
摘要翻译：通信设备可以经由通信设备确定下行链路处理路径的总增益。可以在下行链路处理路径期间，基于通过通信装置支持的多个音频处理装置中的每一个的音频要求的确定来确定总增益。通信设备可操作以基于所确定的音频要求来确定默认混合信号增益，下行链路增益调整和校准增益，并且基于所确定的默认混合信号增益来计算下行链路处理路径的总增益，下行链路增益调整和校准增益。总增益的确定可以基于通信设备中的电话状态的确定。可以在通信设备的校准期间确定校准增益，而可以基于调整步长和/或基于模式的增益索引来计算下行链路增益调整。

9. 发明申请

US20110021241A1 METHOD AND SYSTEM FOR AUDIO SYSTEM VOLUME CONTROL 有权
标题翻译：音频系统音量控制方法与系统
公开(公告)号：US20110021241A1
公开(公告)日：2011-01-27
申请号：US12548758
申请日：2009-08-27
申请人： Prakash Khanduri
发明人： Prakash Khanduri
IPC分类号： H04M1/00 , H03G3/00
CPC分类号： H04M1/6016
摘要： A communication device may determine total gain for a downlink processing path via the communication device. The total gain may be determined based on determination of audio requirements for each of a plurality of audio processing devices supported via the communication device, during the downlink processing path. The communication device is operable to determine a default mix signal gain, a downlink gain adjustment and a calibration gain based on the determined audio requirements, and to calculate the total gain for the downlink processing path based on the determined default mix signal gain, the downlink gain adjustment and the calibration gain. The determination of the total gain may be based on determination of a telephony state in the communication device. The calibration gain may be determined during calibration of the communication device, whereas the downlink gain adjustment may be calculated based on an adjustment step-size and/or a mode based gain index.
摘要翻译：通信设备可以经由通信设备确定下行链路处理路径的总增益。可以在下行链路处理路径期间，基于通过通信装置支持的多个音频处理装置中的每一个的音频要求的确定来确定总增益。通信设备可操作以基于所确定的音频要求来确定默认混合信号增益，下行链路增益调整和校准增益，并且基于所确定的默认混合信号增益来计算下行链路处理路径的总增益，下行链路增益调整和校准增益。总增益的确定可以基于通信设备中的电话状态的确定。可以在通信设备的校准期间确定校准增益，而可以基于调整步长和/或基于模式的增益索引来计算下行链路增益调整。

10. 发明授权

US08559320B2 Method and apparatus for measuring voice quality on a VoIP network 有权
标题翻译：用于测量VoIP网络上语音质量的方法和装置
公开(公告)号：US08559320B2
公开(公告)日：2013-10-15
申请号：US12050991
申请日：2008-03-19
申请人： Prakash Khanduri
发明人： Prakash Khanduri
IPC分类号： H04L12/26
CPC分类号： H04L41/5009 , H04L41/5087 , H04L65/80 , H04M3/2227
摘要： In a method and apparatus for measuring voice quality of an endpoint in a Voice-over-IP (VoIP) network, a plurality of network quality parameters are determined for the endpoint. A voice quality factor for the endpoint is determined in real time based on the network quality parameters.
摘要翻译：在用于测量IP语音（VoIP）网络中的端点的语音质量的方法和装置中，为端点确定多个网络质量参数。基于网络质量参数实时确定端点的语音质量因子。

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