专利快速检索-快速检索全球专利，免费商用专利数据库-IPRDB

51. 发明授权

US09972307B2 System and method for dialog modeling 有权
公开(公告)号：US09972307B2
公开(公告)日：2018-05-15
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： G06F17/22 , G10L15/08 , G10L15/06 , G10L15/183 , G10L15/22 , G10L15/00 , G10L15/04 , G10L25/12 , 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.

52. 发明授权

US09928850B2 Linear predictive analysis apparatus, method, program and recording medium 有权
公开(公告)号：US09928850B2
公开(公告)日：2018-03-27
申请号：US15112318
申请日：2015-01-20
申请人： NIPPON TELEGRAPH AND TELEPHONE CORPORATION
发明人： Yutaka Kamamoto , Takehiro Moriya , Noboru Harada
IPC分类号： G10L19/00 , G10L21/00 , G10L25/12 , G10L25/06 , G10L19/06 , G10L25/90
CPC分类号： G10L25/12 , G10L19/06 , G10L25/06 , G10L25/21 , G10L25/90
摘要： An autocorrelation calculating part calculates autocorrelation Ro(i) from an input signal. A predictive coefficient calculating part performs linear predictive analysis using modified autocorrelation R′o(i) obtained by multiplying the autocorrelation Ro(i) by a coefficient wo(i). Here, it is assumed that a case where, for at least part of each order i, the coefficient wo(i) corresponding to each order i monotonically increases as a value having negative correlation with a fundamental frequency of an input signal in a current frame or a past frame increases and a case where the coefficient wo(i) monotonically decreases as a value having positive correlation with a pitch gain in a current frame or a past frame increases, are included.

53. 发明授权

US09847089B2 Method for reduction of aliasing introduced by spectral envelope adjustment in real-valued filterbanks 有权
公开(公告)号：US09847089B2
公开(公告)日：2017-12-19
申请号：US15392545
申请日：2016-12-28
申请人： Dolby International AB
发明人： Kristofer Kjoerling , Lars Villemoes
IPC分类号： G10L19/00 , G10L19/083 , G10L19/26 , G10L19/02 , G10L25/12 , G10L19/025
CPC分类号： G10L19/0204 , G10L19/00 , G10L19/025 , G10L19/083 , G10L19/12 , G10L19/26 , G10L25/12 , G10L25/18 , G10L25/21 , G10L25/48 , H03H17/0266
摘要： The present invention proposes a new method for improving the performance of a real-valued filterbank based spectral envelope adjuster. By adaptively locking the gain values for adjacent channels dependent on the sign of the channels, as defined in the application, reduced aliasing is achieved. Furthermore, the grouping of the channels during gain-calculation, gives an improved energy estimate of the real valued subband signals in the filterbank.

54. 发明授权

US09685173B2 Method for non-intrusive acoustic parameter estimation 有权
公开(公告)号：US09685173B2
公开(公告)日：2017-06-20
申请号：US14138944
申请日：2013-12-23
申请人： NUANCE COMMUNICATIONS, INC.
发明人： Dushyant Sharma , Patrick Naylor , Pablo Peso Parada
IPC分类号： G10L21/00 , G10L25/12 , G10L25/60
CPC分类号： G10L25/12 , G10L25/60
摘要： A system and method for non-intrusive acoustic parameter estimation is included. The method may include receiving, at a computing device, a first speech signal associated with a particular user. The method may include extracting one or more short-term features from the first speech signal. The method may also include determining one or more statistics of each of the one or more short-term features from the first speech signal. The method may further include classifying the one or more statistics as belonging to one or more acoustic parameter classes.

55. 发明申请

US20170110136A1 METHOD FOR REDUCTION OF ALIASING INTRODUCED BY SPECTRAL ENVELOPE ADJUSTMENT IN REAL-VALUED FILTERBANKS 有权
公开(公告)号：US20170110136A1
公开(公告)日：2017-04-20
申请号：US15392545
申请日：2016-12-28
申请人： Dolby International AB
发明人： Kristofer KJOERLING , Lars VILLEMOES
IPC分类号： G10L19/083 , G10L19/025 , G10L25/12 , G10L19/26 , G10L19/02
CPC分类号： G10L19/0204 , G10L19/00 , G10L19/025 , G10L19/083 , G10L19/12 , G10L19/26 , G10L25/12 , G10L25/18 , G10L25/21 , G10L25/48 , H03H17/0266
摘要： The present invention proposes a new method for improving the performance of a real-valued filterbank based spectral envelope adjuster. By adaptively locking the gain values for adjacent channels dependent on the sign of the channels, as defined in the application, reduced aliasing is achieved. Furthermore, the grouping of the channels during gain-calculation, gives an improved energy estimate of the real valued subband signals in the filterbank.

56. 发明授权

US09626983B2 Temporal gain adjustment based on high-band signal characteristic 有权
公开(公告)号：US09626983B2
公开(公告)日：2017-04-18
申请号：US14731276
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 , G10L19/06 , G10L21/0224 , G10L25/12 , G10L21/038 , G10L19/00 , G10L19/24 , G10L19/02
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.

57. 发明授权

US09583115B2 Temporal gain adjustment based on high-band signal characteristic 有权
标题翻译：基于高频信号特征的时间增益调整
公开(公告)号：US09583115B2
公开(公告)日：2017-02-28
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 , G10L21/038 , G10L19/00 , G10L19/24 , G10L19/02
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.
摘要翻译：本公开提供了用于调整时间增益参数和用于调整线性预测系数的技术。时间增益参数的值可以基于音频信号的合成高频带部分与音频信号的高频带部分的比较。如果高频带部分的较高频率范围的信号特征满足第一阈值，则可以调整时间增益参数。可以基于对LP阶使用第一值的LP增益操作来确定线性预测（LP）增益。 LP增益可以与LP合成滤波器的能级相关联。如果LP增益满足第二阈值，LP阶可以减小。

58. 发明申请

US20170011734A1 METHOD FOR SYSTEM COMBINATION IN AN AUDIO ANALYTICS APPLICATION 审中-公开
标题翻译：用于音频分析应用中的系统组合的方法
公开(公告)号：US20170011734A1
公开(公告)日：2017-01-12
申请号：US14793089
申请日：2015-07-07
申请人： International Business Machines Corporation
发明人： Sriram Ganapathy , Mohamed K. Omar , Robert Ward
IPC分类号： G10L15/00 , G10L25/30 , G10L25/12 , G10L15/02 , G10L15/06
CPC分类号： G10L15/005 , G06F17/275 , G10L15/02 , G10L15/18 , G10L15/32
摘要： Exemplary embodiments of the present invention provide a method of system combination in an audio analytics application including providing a plurality of language identification systems in which each of the language identification systems includes a plurality of probabilities. Each probability is associated with the system's ability to detect a particular language. The method of system combination in the audio analytics application includes receiving data at the language identification systems. The received data is different from data used to train the language identification systems. A confidence measure is determined for each of the language identification systems. The confidence measure identifies which language its system predicts for the received data and combining the language identification systems according to the confidence measures.
摘要翻译：本发明的示例性实施例提供了一种在音频分析应用中的系统组合的方法，包括提供多个语言识别系统，其中每个语言识别系统包括多个概率。每个概率与系统检测特定语言的能力相关联。音频分析应用中的系统组合方法包括在语言识别系统处接收数据。接收的数据与用于训练语言识别系统的数据不同。确定每个语言识别系统的置信度量。置信度量度确定其系统对接收到的数据预测哪种语言，并根据置信度度量结合语言识别系统。

59. 发明授权

US09465867B2 System and method for continuous media segment identification 有权
标题翻译：连续媒体段识别的系统和方法
公开(公告)号：US09465867B2
公开(公告)日：2016-10-11
申请号：US14953994
申请日：2015-11-30
申请人： W. Leo Hoarty
发明人： W. Leo Hoarty
IPC分类号： G10L21/00 , G06F17/30 , G10L25/54 , G10L25/12 , G10L25/06
CPC分类号： H04N21/233 , G06F17/30743 , G06F17/30781 , G06F17/30784 , G06F17/30787 , G06F17/30825 , G06F17/30864 , G10L25/06 , G10L25/12 , G10L25/54 , H04N21/235 , H04N21/236 , H04N21/278 , H04N21/8352
摘要： This invention provides a means to identify unknown media programming using the audio component of said programming. The invention extracts audio information from the media received by consumer electronic devices such as smart TVs and TV set-top boxes then conveys said information to a remote server means which will in turn identify said audio information of unknown identity by way of testing against a database of known audio segment information. The system identifies unknown media programming in real-time such that time-sensitive services may be offered such as interactive television applications providing contextually related information or television advertisement substitution. Other uses include tracking media consumption among many other services.
摘要翻译：本发明提供使用所述节目的音频分量来识别未知媒体节目的方法。本发明从诸如智能电视机和电视机顶盒之类的消费者电子设备接收的媒体中提取音频信息，然后将所述信息传送到远程服务器装置，该远程服务器装置依次通过针对数据库的测试来识别未知身份的所述音频信息的已知音频段信息。该系统实时地识别未知媒体节目，以便可以提供时间敏感的服务，例如提供背景相关信息或电视广告替代的交互式电视应用。其他用途包括跟踪许多其他服务中的媒体消费。

60. 发明申请

US20160134984A1 DETERMINING NOISE AND SOUND POWER LEVEL DIFFERENCES BETWEEN PRIMARY AND REFERENCE CHANNELS 审中-公开
标题翻译：确定主要和参考通道之间的噪音和声压级别差异
公开(公告)号：US20160134984A1
公开(公告)日：2016-05-12
申请号：US14938798
申请日：2015-11-11
申请人： CYPHER, LLC
发明人： Jan S. Erkelens
IPC分类号： H04R29/00 , G10L25/21 , G10L21/0232 , G10L25/12
CPC分类号： G10L21/0232 , G10L25/12 , G10L25/21 , G10L2021/02165 , H04R3/005 , H04R2410/05
摘要： A method for estimating a noise power level difference (NPLD) between a primary microphone and a reference microphone of an audio device includes obtaining primary and reference channels of an audio signal with primary and reference microphones of an audio device and estimating a noise magnitude of the reference channel of the audio signal to provide a noise variance estimate for one or more frequencies. A modelled probability density function (PDF) of a fast Fourier transform (FFT) coefficient of the primary channel of the audio signal is maximized to provide a NPLD between the noise variance estimate of the reference channel and a noise variance estimate of the primary channel. A modelled PDF of an FFT coefficient of the reference channel of the audio signal is maximized to provide a complex speech power level difference (SPLD) coefficient between the speech FFT coefficients of the primary and reference channel. A corrected noise magnitude of the reference channel is then calculated based on the noise variance estimate, the NPLD and the SPLD coefficient.
摘要翻译：用于估计音频设备的主麦克风和参考麦克风之间的噪声功率电平差（NPLD）的方法包括：利用音频设备的主要和参考麦克风来获得音频信号的主要参考信道和参考信道，并估计音频信号的噪声幅度音频信号的参考通道，以提供一个或多个频率的噪声方差估计。将音频信号的主要信道的快速傅里叶变换（FFT）系数的建模概率密度函数（PDF）最大化，以在参考信道的噪声方差估计和主信道的噪声方差估计之间提供NPLD。将音频信号的参考信道的FFT系数的建模PDF最大化，以在主信道和参考信道的语音FFT系数之间提供复合语音功率电平差（SPLD）系数。然后基于噪声方差估计，NPLD和SPLD系数来计算参考信道的校正噪声幅度。

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