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1. 发明授权

US5577117A Methods and apparatus for estimating and adjusting the frequency response of telecommunications channels 失效
标题翻译：用于估计和调整电信频道频率响应的方法和装置
公开(公告)号：US5577117A
公开(公告)日：1996-11-19
申请号：US257129
申请日：1994-06-09
申请人： Rafi Rabipour , Vasu Iyengar
发明人： Rafi Rabipour , Vasu Iyengar
IPC分类号： H04B3/06 , H04R29/00
CPC分类号： H04B3/06 , H04L25/03885
摘要： In methods and apparatus for estimating the frequency response of telecommunications channels, signals carried on the channels are divided into pluralities of signal segments of limited duration, unvoiced signal segments are identified, and spectral components of the unvoiced signal segments are measured. The measured spectral components may be integrated over time to derive time-averaged measurements of the spectral components. The time-averaged measurements may be compared to corresponding components of expected unvoiced signal spectra to improve the accuracy of the frequency response estimates. Filters may be inserted in the channels to adjust the frequency responses, the filters having filter characteristics selected in response to the measured spectral components. The methods and apparatus can be used to compensate for poor bass frequency response in voice channels.
摘要翻译：在用于估计电信信道的频率响应的方法和装置中，在信道上承载的信号被划分成多个有限持续时间的信号段，识别清音信号段，并且测量清音信号段的频谱分量。测量的光谱分量可以随时间积分以导出光谱分量的时间平均测量。可以将时间平均测量值与预期清音信号频谱的相应分量进行比较，以提高频率响应估计的准确度。可以将滤波器插入通道中以调整频率响应，滤波器具有响应于测量的频谱分量而选择的滤波器特性。方法和装置可用于补偿语音信道中的低音低频响应。

2. 发明授权

US5455888A Speech bandwidth extension method and apparatus 失效
标题翻译：语音带宽扩展方法和装置
公开(公告)号：US5455888A
公开(公告)日：1995-10-03
申请号：US985418
申请日：1992-12-04
申请人： Vasu Iyengar , Rafi Rabipour , Paul Mermelstein , Brian R. Shelton
发明人： Vasu Iyengar , Rafi Rabipour , Paul Mermelstein , Brian R. Shelton
IPC分类号： G10L19/00 , G10L19/02 , G10L21/02 , G10L5/06
CPC分类号： G10L21/038 , G10L19/0204 , G10L21/0232
摘要： A speech bandwidth extension method and apparatus analyzes narrowband speech sampled at 8 kHz using LPC analysis to determine its spectral shape and inverse filtering to extract its excitation signal. The excitation signal is interpolated to a sampling rate of 16 kHz and analyzed for pitch control and power level. A white noise generated wideband signal is then filtered to provide a synthesized wideband excitation signal. The narrowband shape is determined and compared to templates in respective vector quantizer codebooks, to select respective highband shape and gain. The synthesized wideband excitation signal is then filtered to provide a highband signal which is, in turn, added to the narrowband signal, interpolated to the 16 kHz sample rate, to produce an artificial wideband signal. The apparatus may be implemented on a digital signal processor chip.
摘要翻译：语音带宽扩展方法和装置分析使用LPC分析以8kHz采样的窄带语音，以确定其频谱形状和反向滤波以提取其激励信号。激励信号内插到16 kHz的采样率，并分析了音调控制和功率电平。然后对产生白噪声的宽带信号进行滤波以提供合成的宽带激励信号。确定窄带形状并将其与各矢量量化器码本中的模板进行比较，以选择各自的高频带形状和增益。然后对合成的宽带激励信号进行滤波以提供高频带信号，该高频带信号又被加到窄带信号中，以16kHz采样率内插，以产生人造宽带信号。该装置可以在数字信号处理器芯片上实现。

3. 发明授权

US07224794B1 Transmitting message playback concurrent with speakerphone operation 有权
标题翻译：与扬声器操作同时发送消息播放
公开(公告)号：US07224794B1
公开(公告)日：2007-05-29
申请号：US09394097
申请日：1999-09-13
申请人： Paul Joseph Davis , Vasu Iyengar , James Charles Popa
发明人： Paul Joseph Davis , Vasu Iyengar , James Charles Popa
IPC分类号： H04M1/00
CPC分类号： H04M1/6033 , H04M1/642 , H04M1/652 , H04M9/082
摘要： A digital telephone answering device having speakerphone capability allows a recorded message to be played back and heard by the far-end party as well as over the local speakerphone by the near-end party during speakerphone operation as if it were a normal receive signal. Moreover, normal speakerphone conversation is possible during this conversational playback mode allowing the far-end party and/or near-end party to break-in over the played back pre-recorded message as desired and be heard by the other party. Both message and conversation signals are preferably (but not necessarily) at similar levels. Also, the message playback at the near end is preferably (but not necessarily) subject to the same speakerphone digital volume control as that received at the far end.
摘要翻译：具有扬声器功能的数字电话应答设备允许在扬声器操作期间由远端方以及近端方在本地扬声器电话上播放并听到所记录的消息，就像它是正常接收信号一样。此外，在该对话回放模式期间，正常的免提电话会话是可能的，允许远端方和/或近端方根据需要插入播放的预先录制的消息，并由另一方听到。消息和会话信号两者都是相似的（但不一定）。此外，近端的消息回放优选（但不一定）经受与远端接收的相同的扬声器数字音量控制。

4. 发明申请

US20070005206A1 Automobile interface 有权
标题翻译：汽车界面
公开(公告)号：US20070005206A1
公开(公告)日：2007-01-04
申请号：US11173736
申请日：2005-07-01
申请人： You Zhang , Jeffery Faneuff , William Hidden , James Hotary , Steven Lee , Vasu Iyengar
发明人： You Zhang , Jeffery Faneuff , William Hidden , James Hotary , Steven Lee , Vasu Iyengar
IPC分类号： G06F17/00
CPC分类号： G10L15/26 , G06F3/16 , G08G1/09675 , G08G1/096775 , G10L15/063 , H04M1/274516 , H04M1/6091
摘要： An automotive system provides an integrated user interface for control and communication functions in an automobile or other type of vehicle. The user interface supports voice enabled interactions, as well as other modes of interaction, such as manual interactions using controls such as dashboard or steering wheel mounted controls. The system also includes interfaces to devices in the vehicle, such as wireless interfaces to mobile devices that are brought into the vehicle. The system also provides interfaces to information sources such as a remote server, for example, for accessing information.
摘要翻译：汽车系统为汽车或其他类型车辆中的控制和通信功能提供集成的用户界面。用户界面支持语音启用的交互，以及其他交互模式，例如使用仪表板或方向盘安装控件等控件的手动交互。该系统还包括与车辆中的设备的接口，诸如与被带入车辆的移动设备的无线接口。该系统还提供诸如远程服务器之类的信息源的接口，例如用于访问信息。

5. 发明授权

US08798278B2 Dynamic gain adjustment based on signal to ambient noise level 有权
标题翻译：基于信号对环境噪声水平的动态增益调整
公开(公告)号：US08798278B2
公开(公告)日：2014-08-05
申请号：US13075536
申请日：2011-03-30
申请人： Steven H. Isabelle , Vasu Iyengar
发明人： Steven H. Isabelle , Vasu Iyengar
IPC分类号： H03G3/20
CPC分类号： G10L21/0208 , H04R1/086 , H04R2410/07
摘要： Methods and apparatuses for deriving a signal-to-noise ratio based at least in part on a measured level of a signal carrying far-end speech, and a measured level of a signal carrying ambient acoustic noise; determining a target gain adjustment based at least in part on the derived signal-to-noise ratio; applying the target gain adjustment to the signal carrying far-end speech to produce a gain-adjusted signal; and providing the gain-adjusted signal for audio output from a communications device.
摘要翻译：至少部分地基于承载远端语音的信号的测量电平以及承载环境声学噪声的信号的测量电平来导出信噪比的方法和装置; 至少部分地基于导出的信噪比来确定目标增益调整; 将目标增益调整应用于携带远端语音的信号以产生增益调整信号; 以及提供用于从通信设备输出音频的增益调整信号。

6. 发明授权

US08620650B2 Rejecting noise with paired microphones 有权
标题翻译：用配对麦克风拒绝噪音
公开(公告)号：US08620650B2
公开(公告)日：2013-12-31
申请号：US13078632
申请日：2011-04-01
申请人： Luke C. Walters , Vasu Iyengar , Martin David Ring
发明人： Luke C. Walters , Vasu Iyengar , Martin David Ring
IPC分类号： G10L21/00
CPC分类号： H04R1/1083 , G10L21/0208 , G10L2021/02165 , H04R1/406 , H04R3/005 , H04R2410/07
摘要： A system for combining signals includes a first microphone generating a first input signal having a first voice component and a first noise component, a second microphone generating a second input signal having a second voice component and a second noise component, a mixing circuit, and an adaptive filter. The mixing circuit applies a first gain having a value α to the first input signal to produce a first scaled signal, applies a second gain having a value 1−α to the second input signal to produce a second scaled signal, and sums the first scaled signal and the second scaled signal to produce a summed signal. The adaptive filter computes an updated value of α to minimize the energy of the summed signal based on the summed signal, the first input signal and the second input signal, and provides the updated value of α to the mixing circuit.
摘要翻译：一种用于组合信号的系统包括产生具有第一语音分量和第一噪声分量的第一输入信号的第一麦克风，产生具有第二声音分量和第二噪声分量的第二输入信号的第二麦克风，混合电路和自适应滤波器。混合电路对第一输入信号施加具有α值的第一增益以产生第一缩放信号，向第二输入信号施加具有值1-α的第二增益以产生第二缩放信号，并将第一缩放信号和第二缩放信号以产生加和信号。自适应滤波器计算更新的α值，以基于求和的信号，第一输入信号和第二输入信号来最小化求和信号的能量，并且将更新的α值提供给混合电路。

7. 发明申请

US20120076321A1 Single Microphone for Noise Rejection and Noise Measurement 审中-公开
标题翻译：单声道噪声抑制和噪声测量
公开(公告)号：US20120076321A1
公开(公告)日：2012-03-29
申请号：US13075732
申请日：2011-03-30
申请人： Martin D. Ring , Vasu Iyengar
发明人： Martin D. Ring , Vasu Iyengar
IPC分类号： H04B15/00 , H04R1/02
CPC分类号： H04R1/1083 , G10L21/02 , H04R1/086 , H04R1/38 , H04R2201/107 , H04R2410/07 , H04R2420/07 , H04R2430/01 , H04R2460/01
摘要： A microphone includes a sensing element having two opposing sides; and a housing including a first acoustic port having an external-facing portion defined in part by a first aperture located on a first housing side and an internal-facing portion defined in part by a first cavity within the housing, the first cavity being coupled to a first side of the element; and a second acoustic port having an external-facing portion defined in part by a second aperture located on the first housing side and an internal-facing portion defined in part by a second cavity within the housing, the second cavity being coupled to a second side of the element. The ports are spaced apart at a distance such that a level of an electrical response by the element to an ambient acoustic noise at 50 dB A-weighted sound pressure level exceeds an internal electrical noise level of the element.
摘要翻译：麦克风包括具有两个相对侧面的传感元件; 以及壳体，其包括第一声学端口，所述第一声学端口具有部分由位于第一壳体侧上的第一孔限定的外部对置部分和由所述壳体内的第一腔部分限定的内部相对部分，所述第一腔体耦合到元素的第一面; 以及第二声学端口，其具有部分由位于第一壳体侧上的第二孔限定的面向外部分和由壳体内的第二腔部分限定的内部相对部分，第二腔体耦合到第二侧面的元素。这些端口间隔开一定距离，使得该元件对于在50dB A加权声压级的环境声学噪声的电响应的水平超过该元件的内部电噪声水平。

8. 发明授权

US06366670B1 Call related information receiver unit 失效
标题翻译：呼叫相关信息接收单元
公开(公告)号：US06366670B1
公开(公告)日：2002-04-02
申请号：US09058203
申请日：1998-04-10
申请人： Paul Joseph Davis , James A. Johanson , Vasu Iyengar , James Charles Popa , Glenn A. Ehrich
发明人： Paul Joseph Davis , James A. Johanson , Vasu Iyengar , James Charles Popa , Glenn A. Ehrich
IPC分类号： H04M100
CPC分类号： H04M1/80 , H04M1/573
摘要： An adjunct Type II caller ID/call waiting (CIDCW) receiver unit is provided which has a greatly improved ability to detect tones and other call related information on a telephone line from a central office while the customer premises equipment is in an off-hook condition. The inventive adjunct CIDCW receiver unit is placed in series between the telephone line from the central office and the customer premises equipment, rather than in parallel with the customer premises equipment as in conventional adjunct CIDCW receiver units. A second telephone line interface (TLI) is included in the adjunct CIDCW receiver unit for connection to the customer premises unit to simulate the impedance of the telephone line. The adjunct CIDCW receiver unit has the ability to disconnect, mute or suppress the microphone signal from the customer premises equipment from being included in the signal received by the call related information receiver portion of the adjunct CIDCW receiver unit.
摘要翻译：提供附件II型呼叫者ID /呼叫等待（CIDCW）接收机单元，其具有在客户驻地设备处于摘机状态时从中心局检测电话线上的音调和其他呼叫相关信息的能力大大提高的能力。本发明的附属CIDCW接收器单元串联在来自中心局的电话线路和客户驻地设备之间，而不是像传统的辅助CIDCW接收器单元那样与客户驻地设备并行放置。附加CIDCW接收器单元中包括第二电话线接口（TLI），用于连接到客户驻地单元以模拟电话线路的阻抗。辅助CIDCW接收器单元具有断开，静音或抑制来自客户驻地设备的麦克风信号被包括在由附加CIDCW接收器单元的呼叫相关信息接收器部分接收的信号中的能力。

9. 发明授权

US5663955A Echo canceller system with shared coefficient memory 失效
标题翻译：具有共享系数存储器的回波消除器系统
公开(公告)号：US5663955A
公开(公告)日：1997-09-02
申请号：US519500
申请日：1995-08-25
申请人： Vasu Iyengar
发明人： Vasu Iyengar
IPC分类号： G10K11/178 , H03H17/02 , H03H21/00 , H04B3/23 , H04M1/60 , H04M9/08
CPC分类号： H04M9/082
摘要： An echo canceller system includes first and second echo cancellers. Each echo canceller includes a foreground filter and an adaptive background filter, with the foreground filter providing the actual echo cancellation and the background filter updating the foreground filter. The echo canceller system also includes send and receive paths, a shared coefficient memory, and a controller for switching the shared coefficient memory between background filters in response to signals along the send and receive paths. The switching includes resetting the shared coefficient memory to prevent any transfer of filter coefficients between the background filters. The background filters operate one at a time, depending on which background filter has access to the shared coefficient memory, while the foreground filters operate simultaneously. The echo canceller system is well-suited for use in loudspeaking telephone sets, with the first echo canceller canceling a line echo through a hybrid transformer, and the second echo canceller canceling an acoustic echo between a loudspeaker and a microphone. The coefficient memory may be switched to the first background filter in response to a near-end signal without a far-end signal (i.e., transmit state), and switched to the second background filter in response to a far-end signal without a near-end signal (i.e., receive state).
摘要翻译：回波消除器系统包括第一和第二回波消除器。每个回波消除器包括前景滤波器和自适应背景滤波器，其中前台滤波器提供实际回波消除，并且背景滤波器更新前景滤波器。回波消除器系统还包括发送和接收路径，共享系数存储器和用于响应于沿着发送和接收路径的信号在后台滤波器之间切换共享系数存储器的控制器。切换包括复位共享系数存储器以防止在后台滤波器之间传送滤波器系数。背景滤镜每次运行一个，这取决于哪个背景滤镜可以访问共享系数存储器，而前景滤镜同时运行。回声消除器系统非常适合用于扬声器电话机，第一回波消除器通过混合变压器取消线路回声，第二回波消除器消除扬声器和麦克风之间的声学回声。系数存储器可以响应于没有远端信号（即发送状态）的近端信号而被切换到第一背景滤波器，并且响应于没有接近的远端信号而切换到第二背景滤波器 -end信号（即接收状态）。

10. 发明申请

US20120253798A1 Rejecting Noise with Paired Microphones 有权
标题翻译：用配对麦克风拒绝噪音
公开(公告)号：US20120253798A1
公开(公告)日：2012-10-04
申请号：US13078632
申请日：2011-04-01
申请人： Luke C. Walters , Vasu Iyengar , Martin David Ring
发明人： Luke C. Walters , Vasu Iyengar , Martin David Ring
IPC分类号： G10L21/02
CPC分类号： H04R1/1083 , G10L21/0208 , G10L2021/02165 , H04R1/406 , H04R3/005 , H04R2410/07
摘要： A system for combining signals includes a first microphone generating a first input signal having a first voice component and a first noise component, a second microphone generating a second input signal having a second voice component and a second noise component, a mixing circuit, and an adaptive filter. The mixing circuit applies a first gain having a value α to the first input signal to produce a first scaled signal, applies a second gain having a value 1−α to the second input signal to produce a second scaled signal, and sums the first scaled signal and the second scaled signal to produce a summed signal. The adaptive filter computes an updated value of α to minimize the energy of the summed signal based on the summed signal, the first input signal and the second input signal, and provides the updated value of α to the mixing circuit.
摘要翻译：一种用于组合信号的系统包括产生具有第一语音分量和第一噪声分量的第一输入信号的第一麦克风，产生具有第二声音分量和第二噪声分量的第二输入信号的第二麦克风，混合电路和自适应滤波器。混合电路对第一输入信号施加具有值α的第一增益以产生第一缩放信号，向第二输入信号施加具有值1-α的第二增益以产生第二缩放信号，并将第一缩放信号和第二缩放信号以产生加和信号。自适应滤波器根据求和信号，第一输入信号和第二输入信号，计算α的更新值，使求和信号的能量最小化，并将更新的α值提供给混合电路。

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