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71. 发明申请

US20160295320A1 AUDIO CAPTURING ENHANCEMENT METHOD AND AUDIO CAPTURING SYSTEM USING THE SAME 有权
标题翻译：使用该方法的音频提取方法和音频捕获系统
公开(公告)号：US20160295320A1
公开(公告)日：2016-10-06
申请号：US14673886
申请日：2015-03-31
申请人： ASUSTeK COMPUTER INC.
发明人： Bhoomek D. Pandya , Nikhil Bhave
IPC分类号： H04R1/40 , G10L25/78 , H03G3/20
CPC分类号： H04R1/406 , G10L21/0208 , G10L25/12 , G10L25/78 , G10L2021/02166 , H03G3/3005 , H04R3/005 , H04R2430/25
摘要： Accordingly, the present disclosure is directed to an audio capturing enhancement method and an audio capturing system using the same method. The audio capturing system includes at least but not limited to two microphones for recording an audio data, an amplifier coupled to the at least two microphones for adjusting the audio data by applying automatic gain control (AGC) in order to generate a gain adjusted data that is within a predefined level, and a processing circuit coupled to the amplifier for calculating a linear predictive coding (LPC) residue of the gain adjusted data, determining from the LPC residue a first source at a first direction relative to the at least two microphones based on time different of arrival (TDOA), and attenuating any source at a second direction that is outside of a predefined direction.
摘要翻译：因此，本公开涉及使用相同方法的音频捕获增强方法和音频捕获系统。音频捕获系统至少包括但不限于用于记录音频数据的两个麦克风，耦合到至少两个麦克风的放大器，用于通过应用自动增益控制（AGC）来调整音频数据，以产生增益调整数据，以及处理电路，其耦合到所述放大器，用于计算所述增益调整数据的线性预测编码（LPC）残差，从所述LPC残差确定相对于所述至少两个麦克风的第一方向上的第一源准时到达（TDOA）不同，并且在预定方向外的第二方向衰减任何源。

72. 发明授权

US09343079B2 Receiver intelligibility enhancement system 有权
标题翻译：接收机可懂度增强系统
公开(公告)号：US09343079B2
公开(公告)日：2016-05-17
申请号：US14468191
申请日：2014-08-25
申请人： Alon Konchitsky
发明人： Alon Konchitsky
IPC分类号： G10L21/0364 , G10L21/0208 , H04R29/00 , H04R3/00 , G10L25/12
CPC分类号： G10L21/0364 , G10L21/0208 , G10L25/12 , H04R3/005
摘要： Embodiments of the invention provide a communication device and methods for enhancing audio signals. A first audio signal buffer and a second audio signal buffer are acquired. Thereafter, the second audio signal is processed based on the linear predictive coding coefficients and gains based on mean noise power of the first audio signal to generate an enhanced second audio signal.
摘要翻译：本发明的实施例提供了一种用于增强音频信号的通信设备和方法。获取第一音频信号缓冲器和第二音频信号缓冲器。此后，基于线性预测编码系数和基于第一音频信号的平均噪声功率的增益处理第二音频信号，以产生增强的第二音频信号。

73. 发明申请

US20160119725A1 PACKET LOSS CONCEALMENT TECHNIQUES FOR PHONE-TO-HEARING-AID STREAMING 有权
标题翻译：用于电话听力助听器的分组丢失隐藏技术
公开(公告)号：US20160119725A1
公开(公告)日：2016-04-28
申请号：US14854704
申请日：2015-09-15
申请人： Frederic Philippe Denis Mustiere , Mohammed Reza Movahedi , Tao Zhang , Jon S. Kindred
发明人： Frederic Philippe Denis Mustiere , Mohammed Reza Movahedi , Tao Zhang , Jon S. Kindred
IPC分类号： H04R25/00 , G10L19/005
CPC分类号： H04R25/554 , G10L19/005 , G10L25/12 , H04R25/305 , H04R25/43 , H04R2225/43
摘要： Embodiments of packet loss concealment for phone-to-hearing-aid streaming are generally described herein. A method for packet loss concealment can include receiving a first frame at a hearing assistance device, determining, at the hearing assistance device, that a second frame was not received within a predetermined time, and determining a first set of sequential samples that match the first frame. The method can include cross-fading the first frame and the first set of sequential samples to create a first cross-faded frame and extrapolating a third frame to replace the second frame using the first set of sequential samples and an autoregressive model.
摘要翻译：这里通常描述用于电话到助听器流的分组丢失隐藏的实施例。一种用于分组丢失隐藏的方法可以包括在听力辅助装置处接收第一帧，在听力辅助装置处确定在预定时间内没有接收到第二帧，并且确定与第一帧匹配的第一组序列样本帧。该方法可以包括使第一帧和第一组顺序样本交叉衰落以创建第一交叉衰落帧并且使用第一组顺序样本和自回归模型来外插第三帧以替换第二帧。

74. 发明授权

US09313573B2 Method and device for microphone selection 有权
标题翻译：麦克风选择方法和设备
公开(公告)号：US09313573B2
公开(公告)日：2016-04-12
申请号：US13980517
申请日：2011-11-16
申请人： Christian Schüldt , Fredric Lindström
发明人： Christian Schüldt , Fredric Lindström
IPC分类号： H04R3/00 , G10L21/02 , G10L21/0264 , G10L25/12 , G10L21/0216
CPC分类号： H04R3/005 , G10L21/02 , G10L21/0264 , G10L25/12 , G10L2021/02166 , H04R2430/03
摘要： The present invention relates to a device, such as an audio communication device, for combining a plurality of microphone signals xn(k) into a single output signal y(k). The device comprises processing means configured to calculate control signals fn(k), and control means configured to select which microphone signal xn(k) or which combination of microphone signals xn(k) to use as output signal y(k) based on said control signals fn(k). To improve the selection, the device comprises linear prediction filters for calculating linear prediction residual signals en(k) from the plurality of microphone signals xn(k), and the processing means is configured to calculate the control signals fn(k) based on said linear prediction residual signals en(k).
摘要翻译：本发明涉及一种用于将多个麦克风信号xn（k）组合成单个输出信号y（k）的装置，诸如音频通信装置。该装置包括被配置为计算控制信号fn（k）的处理装置和被配置为根据所述的麦克风信号xn（k）或哪个麦克风信号xn（k）的组合用作输出信号y（k）的控制装置控制信号fn（k）。为了改善选择，该装置包括用于从多个麦克风信号xn（k）计算线性预测残差信号en（k）的线性预测滤波器，并且处理装置被配置为基于所述多个麦克风信号计算控制信号fn（k）线性预测残差信号en（k）。

75. 发明申请

US20160005414A1 SYSTEM AND METHOD FOR COMPRESSED DOMAIN ESTIMATION OF THE SIGNAL TO NOISE RATIO OF A CODED SPEECH SIGNAL 有权
标题翻译：信号的压缩域估计系统和方法对编码语音信号的噪声比
公开(公告)号：US20160005414A1
公开(公告)日：2016-01-07
申请号：US14322369
申请日：2014-07-02
申请人： Nuance Communications, Inc.
发明人： Jose Lainez , Daniel A. Barreda , Dushyant Sharma , Patrick Naylor , Sridhar Pilli
IPC分类号： G10L19/028 , G10L25/18 , G10L25/90 , G10L19/002
CPC分类号： G10L19/028 , G10L19/002 , G10L21/0216 , G10L25/03 , G10L25/12 , G10L25/18 , G10L25/21 , G10L25/60 , G10L25/69 , G10L25/90 , G10L2019/0002 , G10L2019/0006
摘要： The present disclosure is directed towards a process for estimating the signal to noise ratio of a speech signal. The process may include receiving, at a computing device, a speech signal having a bitstream and a signal-to-noise ratio (“SNR”) associated therewith. The process may further include estimating the SNR directly from the bitstream or using a partial decoder that is configured to extract one or more parameters, the parameters including at least one of a fixed codebook gain, an adaptive codebook gain, a pitch lag, and a line spectral frequency (“LSF”) coefficient.
摘要翻译：本公开涉及用于估计语音信号的信噪比的处理。该过程可以包括在计算设备处接收具有与其相关联的比特流和信噪比（“SNR”）的语音信号。该过程还可以包括直接从比特流估计SNR，或者使用被配置为提取一个或多个参数的部分解码器，该参数包括固定码本增益，自适应码本增益，音调滞后和线谱频率（“LSF”）系数。

76. 发明申请

US20150380006A1 TEMPORAL GAIN ADJUSTMENT BASED ON HIGH-BAND SIGNAL CHARACTERISTIC 有权
公开(公告)号：US20150380006A1
公开(公告)日：2015-12-31
申请号：US14731198
申请日：2015-06-04
申请人： QUALCOMM Incorporated
发明人： Venkatraman S. Atti , Venkatesh Krishnan , Vivek Rajendran , Venkata Subrahmanyam Chandra Sekhar Chebiyyam , Subasingha Shaminda Subasingha
IPC分类号： G10L19/12 , G10L19/032 , G10L21/0224 , G10L19/06 , G10L25/12
CPC分类号： G10L19/12 , G10L19/0204 , G10L19/032 , G10L19/06 , G10L19/24 , G10L21/0224 , G10L21/038 , G10L25/12 , G10L2019/0016
摘要： The present disclosure provides techniques for adjusting a temporal gain parameter and for adjusting linear prediction coefficients. A value of the temporal gain parameter may be based on a comparison of a synthesized high-band portion of an audio signal to a high-band portion of the audio signal. If a signal characteristic of an upper frequency range of the high-band portion satisfies a first threshold, the temporal gain parameter may be adjusted. A linear prediction (LP) gain may be determined based on an LP gain operation that uses a first value for an LP order. The LP gain may be associated with an energy level of an LP synthesis filter. The LP order may be reduced if the LP gain satisfies a second threshold.

77. 发明申请

US20150380001A1 MDCT-BASED COMPLEX PREDICTION STEREO CODING 有权
标题翻译：基于MDCT的复杂预测立体声编码
公开(公告)号：US20150380001A1
公开(公告)日：2015-12-31
申请号：US14793297
申请日：2015-07-07
申请人： DOLBY INTERNATIONAL AB
发明人： Heiko Purnhagen , Pontus Carlsson , Lars Villemoes
IPC分类号： G10L19/008 , H04S3/00 , G10L19/06
CPC分类号： G10L19/008 , G10L19/0212 , G10L19/06 , G10L19/167 , G10L25/12 , H04S3/008 , H04S2400/01
摘要： The invention provides methods and devices for stereo encoding and decoding using complex prediction in the frequency domain. In one embodiment, a decoding method, for obtaining an output stereo signal from an input stereo signal encoded by complex prediction coding and comprising first frequency-domain representations of two input channels, comprises the upmixing steps of: (i) computing a second frequency-domain representation of a first input channel; and (ii) computing an output channel on the basis of the first and second frequency-domain representations of the first input channel, the first frequency-domain representation of the second input channel and a complex prediction coefficient. The upmixing can be suspended responsive to control data.
摘要翻译：本发明提供使用频域中的复数预测的立体声编码和解码的方法和装置。在一个实施例中，一种用于从由复数预测编码编码的输入立体声信号中获得输出立体声信号并包括两个输入声道的第一频域表示的解码方法包括上混合步骤：（i）计算第二频率 - 第一输入通道的域表示; 以及（ii）基于第一输入通道的第一和第二频域表示，第二输入通道的第一频域表示和复数预测系数来计算输出通道。响应于控制数据可以暂停上混。

78. 发明申请

US20150379984A1 SYSTEM AND METHOD FOR DIALOG MODELING 有权
标题翻译：对话建模系统与方法
公开(公告)号：US20150379984A1
公开(公告)日：2015-12-31
申请号：US14845634
申请日：2015-09-04
申请人： AT&T Intellectual Property I, L.P.
发明人： Amanda STENT , Srinivas BANGALORE
IPC分类号： G10L15/06 , G10L15/04 , G10L15/00 , G10L25/12
CPC分类号： G10L15/063 , G06F17/22 , G06F17/2241 , G06F17/227 , G10L15/005 , G10L15/04 , G10L15/08 , G10L15/18 , G10L15/183 , G10L15/22 , G10L25/12 , G10L2015/0638
摘要： Disclosed herein are systems, computer-implemented methods, and computer-readable media for dialog modeling. The method includes receiving spoken dialogs annotated to indicate dialog acts and task/subtask information, parsing the spoken dialogs with a hierarchical, parse-based dialog model which operates incrementally from left to right and which only analyzes a preceding dialog context to generate parsed spoken dialogs, and constructing a functional task structure of the parsed spoken dialogs. The method can further either interpret user utterances with the functional task structure of the parsed spoken dialogs or plan system responses to user utterances with the functional task structure of the parsed spoken dialogs. The parse-based dialog model can be a shift-reduce model, a start-complete model, or a connection path model.
摘要翻译：本文公开了用于对话建模的系统，计算机实现的方法和计算机可读介质。该方法包括接收注释以指示对话行为和任务/子任务信息的口头对话，用从层级，基于解析的对话模型解析口头对话，该对话模型从左向右逐渐操作，并且仅分析前一对话上下文以产生解析的口语对话，并构建解析的语音对话的功能任务结构。该方法还可以用解析的口头对话的功能任务结构或用解析的口语对话的功能性任务结构对用户话语的计划系统响应来解释用户话语。基于分析的对话模型可以是移位减少模型，起始完成模型或连接路径模型。

79. 发明授权

US09129601B2 System and method for dialog modeling 有权
标题翻译：对话建模的系统和方法
公开(公告)号：US09129601B2
公开(公告)日：2015-09-08
申请号：US12324340
申请日：2008-11-26
申请人： Amanda Stent , Srinivas Bangalore
发明人： Amanda Stent , Srinivas Bangalore
IPC分类号： G06F17/22 , G10L15/08 , G10L15/22 , G10L15/183 , G10L15/18
CPC分类号： G10L15/063 , G06F17/22 , G06F17/2241 , G06F17/227 , G10L15/005 , G10L15/04 , G10L15/08 , G10L15/18 , G10L15/183 , G10L15/22 , G10L25/12 , G10L2015/0638
摘要： Disclosed herein are systems, computer-implemented methods, and computer-readable media for dialog modeling. The method includes receiving spoken dialogs annotated to indicate dialog acts and task/subtask information, parsing the spoken dialogs with a hierarchical, parse-based dialog model which operates incrementally from left to right and which only analyzes a preceding dialog context to generate parsed spoken dialogs, and constructing a functional task structure of the parsed spoken dialogs. The method can further either interpret user utterances with the functional task structure of the parsed spoken dialogs or plan system responses to user utterances with the functional task structure of the parsed spoken dialogs. The parse-based dialog model can be a shift-reduce model, a start-complete model, or a connection path model.
摘要翻译：本文公开了用于对话建模的系统，计算机实现的方法和计算机可读介质。该方法包括接收注释以指示对话行为和任务/子任务信息的口头对话，用从层级，基于解析的对话模型解析口头对话，该对话模型从左向右逐渐操作，并且仅分析前一对话上下文以产生解析的口语对话，并构建解析的语音对话的功能任务结构。该方法还可以用解析的口头对话的功能任务结构或用解析的口语对话的功能性任务结构对用户话语的计划系统响应来解释用户话语。基于分析的对话模型可以是移位减少模型，起始完成模型或连接路径模型。

80. 发明申请

US20150081285A1 SPEECH SIGNAL PROCESSING APPARATUS AND METHOD FOR ENHANCING SPEECH INTELLIGIBILITY 有权
标题翻译：语音信号处理装置和提高语音智能的方法
公开(公告)号：US20150081285A1
公开(公告)日：2015-03-19
申请号：US14328186
申请日：2014-07-10
申请人： SAMSUNG ELECTRONICS CO., LTD. , YONSEI UNIVERSITY WONJU INDUSTRY-ACADEMIC COOPERATION FOUNDATION
发明人： Jun Il SOHN , Yun Seo KU , Dong Wook KIM , Young Cheol PARK
IPC分类号： G10L25/93
CPC分类号： G10L25/93 , G10L19/04 , G10L21/02 , G10L21/0224 , G10L21/0232 , G10L25/12 , G10L25/15
摘要： A speech signal processing apparatus and a speech signal processing method for enhancing speech intelligibility are provided. The speech signal processing apparatus includes an input signal gain determiner to determine a gain of an input signal based on a harmonic characteristic of a voiced speech, a voiced speech output unit to output a voiced speech in which a harmonic component is preserved by applying the gain to the input signal, a linear predictive coefficient determiner to determine a linear predictive coefficient based on the voiced speech, and an unvoiced speech preserver to preserve an unvoiced speech of the input signal based on the linear predictive coefficient.
摘要翻译：提供了一种用于增强语音清晰度的语音信号处理装置和语音信号处理方法。语音信号处理装置包括：输入信号增益确定器，用于基于有声语音的谐波特性来确定输入信号的增益;有声语音输出单元，用于输出通过应用增益来保存谐波分量的有声语音基于所述有声语音来确定线性预测系数的线性预测系数确定器和基于所述线性预测系数来保留所述输入信号的无声语音的清音语音保持器。

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