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1. 发明授权

US07711107B1 Perceptual masking of residual echo 有权
标题翻译：残余回声的感知掩蔽
公开(公告)号：US07711107B1
公开(公告)日：2010-05-04
申请号：US11129450
申请日：2005-05-12
申请人： Carlo Murgia , Jeffrey D. Klein , Adil Benyassine , Eyal Shlomot , Yang Gao
发明人： Carlo Murgia , Jeffrey D. Klein , Adil Benyassine , Eyal Shlomot , Yang Gao
IPC分类号： H04M9/08
CPC分类号： H04B3/234
摘要： A method of masking a residual echo signal by an echo canceller is provided. The method comprises receiving a far-end signal, adjusting filter coefficients of an adaptive filter in response to the far-end signal, generating an echo model signal based on the far-end signal using the adaptive filter, receiving a near-end signal, subtracting the echo model signal from the near-end signal to generate an output signal, defining a spectral mask based on the near-end signal, wherein the spectral mask is indicative of near-end spectral peaks and near-end spectral valleys, de-emphasizing the output signal in spectral regions of the near-end spectral peaks, and emphasizing the output signal in spectral regions of the near-end spectral valleys, wherein the de-emphasizing occurs during filter coefficients determination for the adaptive filter. A weighted filter may perform the de-emphasizing and the emphasizing operations, where the weighted filter uses medium term spectral characteristics of the near-end signal.
摘要翻译：提供了一种通过回波消除器掩蔽残留回波信号的方法。该方法包括接收远端信号，响应于远端信号调整自适应滤波器的滤波器系数，使用自适应滤波器基于远端信号生成回波模型信号，接收近端信号，从近端信号减去回波模型信号以产生输出信号，基于近端信号定义频谱屏蔽，其中频谱掩模表示近端谱峰和近端谱谷，强调近端光谱峰值的光谱区域中的输出信号，并且强调近端光谱谷的光谱区域中的输出信号，其中在自适应滤波器的滤波器系数确定期间发生去加重。加权滤波器可以执行去强调和强调操作，其中加权滤波器使用近端信号的中期频谱特性。

2. 发明授权

US07558729B1 Music detection for enhancing echo cancellation and speech coding 有权
标题翻译：用于增强回声消除和语音编码的音乐检测
公开(公告)号：US07558729B1
公开(公告)日：2009-07-07
申请号：US11084392
申请日：2005-03-17
申请人： Adil Benyassine , Yang Gao , Carlo Murgia , Eyal Shlomot
发明人： Adil Benyassine , Yang Gao , Carlo Murgia , Eyal Shlomot
IPC分类号： G10L21/02 , G10L19/14 , G10L15/20 , H04B3/20 , H04M9/08
CPC分类号： G10L25/48 , G10L25/78
摘要： A method of using music detection to enhance an operation of an echo canceller is provided, wherein the echo canceller includes an adaptive filter and a nonlinear processor. The method comprises receiving an input signal including an echo signal by the echo canceller from a near end device, filtering the input signal using the adaptive filter to eliminate linear components of the echo signal in the input signal and generate an error signal, analyzing the error signal using a music detector to determine existence of a music signal in the error signal, bypassing the nonlinear processor if the analyzing determines the music signal exists in the error signal, and eliminating nonlinear components of the echo signal from the error signal using the nonlinear processor if the analyzing determines the music signal does not exist in the error signal.
摘要翻译：提供了一种使用音乐检测来增强回声消除器的操作的方法，其中回波消除器包括自适应滤波器和非线性处理器。该方法包括从近端设备接收包括回声消除器的回波信号的输入信号，使用自适应滤波器对输入信号进行滤波，以消除输入信号中的回波信号的线性分量并产生误差信号，分析误差使用音乐检测器的信号来确定误差信号中的音乐信号的存在，如果分析确定音乐信号存在于误差信号中，则绕过非线性处理器，并且使用非线性处理器从误差信号中消除回波信号的非线性分量如果分析确定音乐信号不存在于错误信号中。

3. 发明授权

US07787597B1 Automated tools for testing echo cancellers using natural speech excitations 有权
标题翻译：使用自然语音激励测试回波消除器的自动化工具
公开(公告)号：US07787597B1
公开(公告)日：2010-08-31
申请号：US11318373
申请日：2005-12-22
申请人： Eyal Shlomot , Jeffrey D. Klein , Carlo Murgia , Adil Benyassine
发明人： Eyal Shlomot , Jeffrey D. Klein , Carlo Murgia , Adil Benyassine
IPC分类号： H04M1/24
CPC分类号： H04M9/082
摘要： There are provided methods and systems for automated testing of echo cancellers using natural speech excitations and evaluating an echo canceller by transmitting a first signal to the echo canceller, wherein the first signal includes a first speech signal and a first marker signal, and wherein the first marker signal is transmitted a first period of time after the first speech signal is transmitted; receiving a second signal from the echo canceller, wherein the second signal includes a second speech signal and a second marker signal; aligning the first speech signal and the second speech signal using the first marker signal and the second marker signal; determining a choppiness of the second speech signal, when a non-linear processor of the echo canceller is on; and determining an audible echo, when a non-linear processor of the echo canceller is off.
摘要翻译：提供了使用自然语音激励自动测试回波消除器的方法和系统，并且通过向回声消除器发送第一信号来评估回波消除器，其中第一信号包括第一语音信号和第一标记信号，并且其中第一在发送第一语音信号之后的第一时间段发送标记信号; 从所述回声消除器接收第二信号，其中所述第二信号包括第二语音信号和第二标记信号; 使用第一标记信号和第二标记信号对第一语音信号和第二语音信号; 当所述回波消除器的非线性处理器处于接通状态时，确定所述第二语音信号的复杂度; 并且当回波消除器的非线性处理器关闭时，确定可听回音。

4. 发明授权

US07873513B2 Speech transcoding in GSM networks 有权
标题翻译： GSM网络中的语音转码
公开(公告)号：US07873513B2
公开(公告)日：2011-01-18
申请号：US11825424
申请日：2007-07-06
申请人： Carlo Murgia , Yang Gao , Aruna Vittal , Eyal Shlomot
发明人： Carlo Murgia , Yang Gao , Aruna Vittal , Eyal Shlomot
IPC分类号： G10L19/12
CPC分类号： G10L19/173
摘要： There is provided a method of transcoding an Enhance Full Rate (EFR) 12.2 Kbps encoded frame into an Adaptive Multi-Rate (AMR) 12.2 Kbps encoded frame, where the method comprises receiving the EFR 12.2 Kbps encoded frame from a first codec; determining if the EFR 12.2 Kbps encoded frame is a Silence Insertion Descriptor (SID) frame; if the EFR 12.2 Kbps encoded frame is determined to be the SID frame, the method further comprises transcoding the EFR SID frame. There is also provided a method of transcoding an EFR 12.2 Kbps encoded frame into an AMR 12.2 Kbps encoded frame, where the method comprises receiving the AMR 12.2 Kbps encoded frame from a first codec; determining if the AMR 12.2 Kbps encoded frame is an SID frame; if the AMR 12.2 Kbps encoded frame is determined to be the SID frame, the method further comprises transcoding the AMR SID frame.
摘要翻译：提供了一种将增强全速率（EFR）12.2Kbps编码的帧转码为自适应多速率（AMR）12.2Kbps编码帧的方法，其中该方法包括从第一编解码器接收EFR12.2Kbps编码帧; 确定EFR 12.2Kbps编码帧是否是静音插入描述符（SID）帧; 如果EFR12.2Kbps编码帧被确定为SID帧，则该方法还包括对EFR SID帧进行代码转换。还提供了将EFR12.2Kbps编码帧转码为AMR 12.2Kbps编码帧的方法，其中该方法包括从第一编解码器接收AMR 12.2Kbps编码帧; 确定AMR 12.2Kbps编码帧是否是SID帧; 如果AMR 12.2Kbps编码帧被确定为SID帧，则该方法还包括对AMR SID帧进行代码转换。

5. 发明申请

US20090012784A1 Speech transcoding in GSM networks 有权
标题翻译： GSM网络中的语音转码
公开(公告)号：US20090012784A1
公开(公告)日：2009-01-08
申请号：US11825424
申请日：2007-07-06
申请人： Carlo Murgia , Yang Gao , Aruna Vittal , Eyal Shlomot
发明人： Carlo Murgia , Yang Gao , Aruna Vittal , Eyal Shlomot
IPC分类号： G10L19/00
CPC分类号： G10L19/173
摘要： There is provided a method of transcoding an Enhance Full Rate (EFR) 12.2 Kbps encoded frame into an Adaptive Multi-Rate (AMR) 12.2 Kbps encoded frame, where the method comprises receiving the EFR 12.2 Kbps encoded frame from a first codec; determining if the EFR 12.2 Kbps encoded frame is a Silence Insertion Descriptor (SID) frame; if the EFR 12.2 Kbps encoded frame is determined to be the SID frame, the method further comprises transcoding the EFR SID frame. There is also provided a method of transcoding an EFR 12.2 Kbps encoded frame into an AMR 12.2 Kbps encoded frame, where the method comprises receiving the AMR 12.2 Kbps encoded frame from a first codec; determining if the AMR 12.2 Kbps encoded frame is an SID frame; if the AMR 12.2 Kbps encoded frame is determined to be the SID frame, the method further comprises transcoding the AMR SID frame.
摘要翻译：提供了一种将增强全速率（EFR）12.2Kbps编码的帧转码为自适应多速率（AMR）12.2Kbps编码帧的方法，其中该方法包括从第一编解码器接收EFR12.2Kbps编码帧; 确定EFR 12.2Kbps编码帧是否是静音插入描述符（SID）帧; 如果EFR12.2Kbps编码帧被确定为SID帧，则该方法还包括对EFR SID帧进行代码转换。还提供了将EFR12.2Kbps编码帧转码为AMR 12.2Kbps编码帧的方法，其中该方法包括从第一编解码器接收AMR 12.2Kbps编码帧; 确定AMR 12.2Kbps编码帧是否是SID帧; 如果AMR 12.2Kbps编码帧被确定为SID帧，则该方法还包括对AMR SID帧进行代码转换。

6. 发明授权

US07613291B1 Echo path change detection using dual sparse filtering 有权
标题翻译：使用双稀疏滤波的回波路径变化检测
公开(公告)号：US07613291B1
公开(公告)日：2009-11-03
申请号：US11201637
申请日：2005-08-10
申请人： Adil Benyassine , Carlo Murgia
发明人： Adil Benyassine , Carlo Murgia
IPC分类号： H04M9/08
CPC分类号： H04B3/237 , H04B3/234
摘要： There is provided a method for use by an echo canceller to detect an echo path change and adjust to the echo path change. The method comprises determining a first bulk delay using a SPARSE foreground adaptive filter; configuring the foreground adaptive filter to an open-loop mode; canceling the echo signal based on the first bulk delay using the foreground adaptive filter; determining a second bulk delay of the echo signal using a SPARSE background adaptive filter; configuring the foreground adaptive filter to a closed-loop mode and continuing to cancel the echo signal based on the first bulk delay; configuring the background adaptive filter to the open-loop mode; measuring echo cancellation performance of the foreground adaptive filter and the background adaptive filter; and changing parameters of the foreground adaptive filter if the echo cancellation performance of the background adaptive filter is better than the foreground adaptive filter.
摘要翻译：提供了一种由回波消除器使用的方法来检测回波路径变化并适应回波路径变化。该方法包括使用SPARSE前景自适应滤波器来确定第一批量延迟; 将前景自适应滤波器配置为开环模式; 使用前景自适应滤波器基于第一批量延迟来消除回波信号; 使用SPARSE背景自适应滤波器确定回波信号的第二批量延迟; 将前景自适应滤波器配置为闭环模式，并且基于第一批量延迟继续抵消回波信号; 将背景自适应滤波器配置为开环模式; 测量前景自适应滤波器和背景自适应滤波器的回波消除性能; 如果背景自适应滤波器的回波消除性能优于前景自适应滤波器，则改变前景自适应滤波器的参数。

7. 发明申请

US20160066087A1 JOINT NOISE SUPPRESSION AND ACOUSTIC ECHO CANCELLATION 审中-公开
标题翻译：联合噪声抑制和声音消除
公开(公告)号：US20160066087A1
公开(公告)日：2016-03-03
申请号：US14167920
申请日：2014-01-29
申请人： Ludger Solbach , Carlo Murgia
发明人： Ludger Solbach , Carlo Murgia
IPC分类号： H04R3/00
CPC分类号： G10L21/0232 , G10L25/84 , G10L2021/02082 , G10L2021/02166 , H04R3/005 , H04R2410/01 , H04R2410/05
摘要： Systems and methods for joint noise and echo suppression using noise and echo subtraction processing are provided. The noise subtraction processing comprises receiving at least a primary acoustic signal, a secondary acoustic signal, and an echo reference acoustic signal. A desired signal component may be calculated and subtracted from the secondary acoustic signal to obtain a noise component signal. An adjusting coefficient for the noise component signal and an adjusting coefficient for the echo reference signal may be determined and applied. The noise component signal and echo reference signal may be subtracted from the primary acoustic signal to generate a noise and echo subtracted signal. An additional echo cancellation in the noise and echo subtracted signal may be carried out by applying a nonlinear processor. The non-linear processor is driven by at least a ratio of the noise and echo subtracted signal energy and the primary acoustic signal energy.
摘要翻译：提供了使用噪声和回波减法处理的联合噪声和回波抑制的系统和方法。噪声减除处理包括至少接收主声信号，次声信号和回波参考声信号。可以计算期望的信号分量并从次声信号中减去以获得噪声分量信号。可以确定并应用噪声分量信号的调整系数和回波参考信号的调整系数。可以从主声信号中减去噪声分量信号和回波参考信号，以产生噪声和回波减去的信号。可以通过应用非线性处理器来执行噪声和回波减去信号中的附加回声消除。非线性处理器由至少噪声和回波相减信号能量与主声信号能量的比率驱动。

8. 发明申请

US20110182436A1 Adaptive Noise Reduction Using Level Cues 有权
标题翻译：使用级别线索自适应降噪
公开(公告)号：US20110182436A1
公开(公告)日：2011-07-28
申请号：US12693998
申请日：2010-01-26
申请人： Carlo Murgia , Carlo Avendano , Karim Younes , Mark Every , Ye Jiang
发明人： Carlo Murgia , Carlo Avendano , Karim Younes , Mark Every , Ye Jiang
IPC分类号： G10K11/16
CPC分类号： G10K11/16 , H04R3/005
摘要： An array of microphones utilizes two sets of two microphones for noise suppression. A primary microphone and secondary microphone of the three microphones may be positioned closely spaced to each other to provide acoustic signals used to achieve noise cancellation. A tertiary microphone may be spaced with respect to either the primary microphone or the secondary microphone in a spread-microphone configuration for deriving level cues from audio signals provided by tertiary and the primary or secondary microphone. Signals from two microphones may be used rather than three microphones. The level cues are expressed via an inter-microphone level difference (ILD) which is used to determine one or more cluster tracking control signals. The ILD based cluster tracking signals are used to control the adaptation of null-processing noise cancellation modules. A noise cancelled primary acoustic signal and ILD based cluster tracking control signals are used during post filtering to adaptively generate a mask to be applied against a speech estimate signal.
摘要翻译：一组麦克风使用两组麦克风进行噪声抑制。三个麦克风的主麦克风和次麦克风可以彼此紧密地定位，以提供用于实现噪声消除的声信号。第三麦克风可以在扩展麦克风配置中相对于主麦克风或辅助麦克风间隔开，以从由三级麦克风和主麦克风或辅助麦克风提供的音频信号导出电平提示。可以使用来自两个麦克风的信号而不是三个麦克风。通过用于确定一个或多个集群跟踪控制信号的麦克风间级差（ILD）来表示电平提示。基于ILD的群集跟踪信号用于控制空处理噪声消除模块的适应。在后滤波期间使用噪声消除的主声信号和基于ILD的群集跟踪控制信号来自适应地生成针对语音估计信号应用的掩码。

9. 发明申请

US20090323982A1 System and method for providing noise suppression utilizing null processing noise subtraction 有权
标题翻译：利用零处理噪声减法提供噪声抑制的系统和方法
公开(公告)号：US20090323982A1
公开(公告)日：2009-12-31
申请号：US12215980
申请日：2008-06-30
申请人： Ludger Solbach , Carlo Murgia
发明人： Ludger Solbach , Carlo Murgia
IPC分类号： H04B15/04 , H04B15/00
CPC分类号： G10L21/0232 , G10L21/0308 , G10L2021/02166 , H04R3/005 , H04R2410/01 , H04R2410/05
摘要： Systems and methods for noise suppression using noise subtraction processing are provided. The noise subtraction processing comprises receiving at least a primary and a secondary acoustic signal. A desired signal component may be calculated and subtracted from the secondary acoustic signal to obtaining a noise component signal. A determination may be made of a reference energy ratio and a prediction energy ratio. A determination may be made as to whether to adjust the noise component signal based partially on the reference energy ratio and partially on the prediction energy ratio. The noise component signal may be adjusted or frozen based on the determination. The noise component signal may then be removed from the primary acoustic signal to generate a noise subtracted signal which may be outputted.
摘要翻译：提供了使用噪声减除处理进行噪声抑制的系统和方法。噪声减除处理包括接收至少一个初级和次级声学信号。可以计算期望的信号分量并从次声信号中减去以获得噪声分量信号。可以确定参考能量比和预测能量比。可以基于参考能量比部分地基于预测能量比来确定是否调整噪声分量信号。可以基于该确定来调整或冻结噪声分量信号。然后可以从主声信号中去除噪声分量信号，以产生可输出的噪声相减信号。

10. 发明授权

US06369722B1 Coding, decoding and transcoding methods 有权
标题翻译：编码，解码和转码方法
公开(公告)号：US06369722B1
公开(公告)日：2002-04-09
申请号：US09527633
申请日：2000-03-17
申请人： Carlo Murgia , Gaël Richard , Philipe Lockwood
发明人： Carlo Murgia , Gaël Richard , Philipe Lockwood
IPC分类号： H03M700
CPC分类号： G10L19/12 , H03M7/30 , H03M7/40 , H03M7/42
摘要： On the basis of a portion of the signal, a coder selects a parameter vector belonging to a reference library containing 2Q vectors each designated by an address of Q bits. This coder, or a transcoder located downstream, forms a digital data stream (&PHgr;) containing an index deduced from the address of the vector selected. For each rate value corresponding to an integer p≧0, the index contained in the digital data stream is formed of Q−p bits which, completed by p bits of predetermined positions, define indices of Q bits representing a group of 2p addresses including that of the vector selected from the reference library. The decoder receiving the digital data stream, or a transcoder located upstream, is capable of reconstructing appropriate parameter vectors on the basis of the truncated index. Thus very fine steps are obtained in the adjustment of the transmission rate.
摘要翻译：基于信号的一部分，编码器选择属于包含由Q位地址指定的2Q个矢量的参考库的参数矢量。该编码器或位于下游的代码转换器形成包含从所选向量的地址推导的索引的数字数据流（＆amp; PHgr）。对于对应于整数p> = 0的每个速率值，包含在数字数据流中的索引由Qp位形成，其由预定位置的p位完成，定义表示一组2p地址的Q位的索引，包括从参考库中选择的向量。接收数字数据流的解码器或位于上游的代码转换器能够基于截断的索引重建适当的参数向量。因此，在传输速率的调整中获得非常好的步骤。

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