专利快速检索-快速检索全球专利，免费商用专利数据库-IPRDB

11. 发明授权

EP2828852B1 POST-PROCESSING GAINS FOR SIGNAL ENHANCEMENT 有权
标题翻译：后处理增益改善信号
公开(公告)号：EP2828852B1
公开(公告)日：2016-03-16
申请号：EP13714812.8
申请日：2013-03-21
申请人： Dolby Laboratories Licensing Corporation
发明人： SUN, Xuejing , DICKINS, Glenn N.
IPC分类号： G10L21/0316 , H03G3/32 , G10K11/16
CPC分类号： G10L21/0364 , G10K11/16 , G10L21/0224 , G10L21/0316 , G10L21/034 , G10L25/78 , H03G3/301 , H03G3/32
摘要： A method, an apparatus, and logic to post-process raw gains determined by input processing to generate post-processed gains, comprising using one or both of delta gain smoothing and decision-directed gain smoothing. The delta gain smoothing comprises applying a smoothing filter to the raw gain with a smoothing factor that depends on the gain delta: the absolute value of the difference between the raw gain for the current frame and the post-processed gain for a previous frame. The decision-directed gain smoothing comprises converting the raw gain to a signal-to-noise ratio, applying a smoothing filter with a smoothing factor to the signal-to-noise ratio to calculate a smoothed signal-to-noise ratio, and converting the smoothed signal-to-noise ratio to determine the second smoothed gain, with smoothing factor possibly dependent on the gain delta.

12. 发明公开

EP2779162A3 Removal of audio noise 有权
标题翻译： Entfernung vonAudiogeräuschen
公开(公告)号：EP2779162A3
公开(公告)日：2015-10-07
申请号：EP14159149.5
申请日：2014-03-12
申请人： Comcast Cable Communications, LLC
发明人： Des Jardins, George Thomas
IPC分类号： G10L21/0224 , G10L21/0208 , G10L19/018
CPC分类号： G10L21/0308 , G10L19/018 , G10L21/0208 , G10L21/0224 , G10L25/84
摘要： A system for removing noise from an audio signal is described. For example, noise caused by content playing in the background during a voice command or phone call may be removed from the audio signal representing the voice command or phone call. By removing noise, the signal to noise ratio of the audio signal may be improved.
摘要翻译：描述用于从音频信号中去除噪声的系统。例如，可以从表示语音命令或电话呼叫的音频信号中去除在语音命令或电话呼叫期间在后台播放的内容引起的噪声。通过消除噪声，可以提高音频信号的信噪比。

13. 发明公开

EP2894449A1 VOICE RECOGNITION METHOD, VOICE RECOGNITION DEVICE, AND ELECTRONIC DEVICE 审中-公开
标题翻译： SPRACHERKENNUNGSVERFAHREN，SPRACHERKENNUNGSVORRICHTUNG ELEKTRONISCHE VORRICHTUNG
公开(公告)号：EP2894449A1
公开(公告)日：2015-07-15
申请号：EP14812390.4
申请日：2014-08-01
申请人： Huawei Technologies Co., Ltd.
发明人： ZHOU, Junyang
IPC分类号： G01L15/00
CPC分类号： G10L15/20 , G10L21/02 , G10L21/0216 , G10L21/0224 , H04R3/00
摘要： The present invention discloses a voice recognition method, a voice recognition device, and an electronic device. In this method, first determining is performed by using a sample environment corresponding to a detection voice and a previous environment type, so as to output a corresponding voice correction instruction to a voice engine; then, a to-be-recognized voice is input to the voice engine and a noise type detection engine at the same time, and the voice engine corrects the to-be-recognized voice by using the voice correction instruction, so that quality of an original voice is not impaired by noise processing, and a corresponding initial recognition result is output; the noise type detection engine determines a current environment type by using the to-be-recognized voice and a voice training sample under a different environment; finally, confidence of the initial recognition result is adjusted by using the current environment type, so as to ensure that a recognition effect of a finally output voice recognition result can provide good user experience for a user under a current environment.
摘要翻译：本发明公开了一种语音识别方法，语音识别装置和电子装置。在该方法中，通过使用与检测语音和先前环境类型对应的样本环境来执行第一确定，以便将语音校正指令输出到语音引擎; 然后，将被识别的语音同时输入到语音引擎和噪声类型检测引擎，并且语音引擎通过使用语音校正指令来校正被识别的语音，使得质量原始声音不受噪声处理的影响，并输出相应的初始识别结果; 噪声型检测引擎通过在不同环境下使用被识别的语音和语音训练样本来确定当前的环境类型; 最后，通过使用当前环境类型来调整初始识别结果的置信度，以确保最终输出的语音识别结果的识别效果可以在当前环境下为用户提供良好的用户体验。

14. 发明公开

EP2761762A1 PREDICTION-BASED FM STEREO RADIO NOISE REDUCTION 有权
标题翻译：调频立体声收音机降噪ON预测的基础
公开(公告)号：EP2761762A1
公开(公告)日：2014-08-06
申请号：EP12772270.0
申请日：2012-10-01
申请人： Dolby International AB
发明人： PURNHAGEN, Heiko , SEHLSTROM, Leif , ENGDEGARD, Jonas
IPC分类号： H04B1/16 , G10L19/008
CPC分类号： H04B1/10 , G10L19/005 , G10L19/008 , G10L21/0224 , G10L21/0264 , H04B1/1676
摘要： The present document relates to audio signal processing, in particular to an apparatus and a corresponding method for improving the audio signal of an FM stereo radio receiver. In particular, the present document relates to a method and system for reducing the noise of a received FM stereo radio signal. An apparatus (2) configured to reduce noise of a received multi-channel FM radio signal is described, wherein the received multi-channel FM radio signal is representable as a received mid signal and a received side signal. The apparatus (2) comprises a parameter determination unit (77) configured to determine one or more parameters indicative of a correlation and/or decorrelation between the received mid signal and the received side signal; and a noise reduction unit (79) configured to generate a noise-reduced side signal from the received mid signal and not from the received side signal using the one or more parameters.

15. 发明公开

EP2622829A1 FINE/COARSE GAIN ADJUSTMENT 无效
标题翻译： FEINE / GROBEVERSTÄRKUNGSEINSTELLUNG
公开(公告)号：EP2622829A1
公开(公告)日：2013-08-07
申请号：EP11761455.2
申请日：2011-09-16
申请人： Bose Corporation
发明人： IYENGAR, Vasu
IPC分类号： H04M1/60
CPC分类号： H04M1/60 , G10L21/0224 , H03G3/32 , H04M1/6066
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, including determining a coarse gain adjustment and a fine gain adjustment, the target gain adjustment corresponding to a combination of the coarse gain adjustment and the fine gain adjustment, applying the coarse gain adjustment to the signal carrying far-end speech using first gain adjustment circuitry to produce a first gain-adjusted signal, applying the fine gain adjustment to the signal carrying far-end speech using second gain adjustment circuitry to produce a second gain-adjusted signal, and providing a result of combining the first gain-adjusted signal and the second gain-adjusted signal to audio output from a wireless 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，至少基于至少部分地基于导出的信噪比，包括确定粗调增益调整和精细增益调整，对应于粗增益调整和精细增益调整的组合的目标增益调整，将粗增益调整应用于使用第一增益调整电路携带远端语音的信号，以产生第一增益调整信号，使用第二增益调整电路对携带远端语音的信号应用精细增益调整，以产生第二增益调整信号，将第一增益调整信号和第二增益调整信号组合成从无线通信设备输出的音频的结果。

16. 发明授权

EP3252765B1 NOISE SUPPRESSION IN A VOICE SIGNAL 有权
公开(公告)号：EP3252765B1
公开(公告)日：2018-12-12
申请号：EP17173212.6
申请日：2017-05-29
申请人： Fujitsu Limited
发明人： Matsuo, Naoshi
IPC分类号： G10L21/0208 , H03G3/32 , H03G3/34 , G10L21/0216 , G10L21/0316 , G10L21/0224
CPC分类号： G10L21/0264 , G10L21/0208 , G10L21/0224 , G10L21/0316 , G10L25/06 , G10L25/21 , G10L2021/02082 , G10L2021/02087 , G10L2021/02165 , H03G3/32 , H04R3/005 , H04R3/04
摘要： A noise suppression device includes: an adaptive filter unit that suppresses, using an adaptive filter, a noise component contained in a voice signal generated from a voice captured by a voice input unit to generate a corrected voice signal; a noise generation detection unit that detects timing of generation of the noise component in the voice signal; and a period suppression unit that suppresses the corrected voice signal during a predetermined period of time after the timing of the generation of the noise component.

17. 发明公开

EP3392883A1 METHOD FOR PROCESSING AN INPUT AUDIO SIGNAL AND CORRESPONDING ELECTRONIC DEVICE, NON-TRANSITORY COMPUTER READABLE PROGRAM PRODUCT AND COMPUTER READABLE STORAGE MEDIUM 审中-公开
公开(公告)号：EP3392883A1
公开(公告)日：2018-10-24
申请号：EP18165900.4
申请日：2018-04-05
申请人： THOMSON LICENSING
发明人： PAREKH, Sanjeel , OZEROV, Alexey , DUONG, Quang Khanh Ngoc , RICHARD, Gaël , ESSID, Slim , PEREZ, Patrick
IPC分类号： G10L21/0272 , G10L25/57
CPC分类号： G10L21/028 , G10L21/0224 , G10L21/0232 , G10L21/0272 , G10L25/57
摘要： The present disclosure relates to a method for processing an input signal comprising an audio component and to the corresponding electronic device, non-transitory computer readable program product and computer readable storage medium.
According to an embodiment of the present disclosure, the method comprises:
• obtaining a set of time parameters from a time frequency transformation of the audio component of the input signal, said audio component being a mixture of audio signals comprising at least one first audio signal of a first audio source;
• determining at least one motion feature of said first audio source from a visual sequence corresponding to the first audio signal;
• obtaining a weight vector of the set of time parameters based on the motion feature; and
• determining a time frequency transformation of the first audio signal based on the weight vector.

18. 发明公开

EP3291234A1 VERFAHREN ZUM BEURTEILEN EINER QUALITÄT EINES STIMMEINSATZES EINES SPRECHENDEN 有权
标题翻译：评估讲话语音互换质量的方法
公开(公告)号：EP3291234A1
公开(公告)日：2018-03-07
申请号：EP16186498.8
申请日：2016-08-31
申请人： Digithep GmbH
发明人： Hönig, Florian
IPC分类号： G10L25/60 , G10L25/66 , G10L25/21
CPC分类号： G10L21/0205 , G10L15/20 , G10L21/0224 , G10L21/0232 , G10L25/21 , G10L25/60 , G10L25/66
摘要： Offenbart wird ein Verfahren zum Beurteilen einer Qualität eines Stimmeinsatzes eines Sprechenden, wobei ein akustisches Sprachsignal des Sprechenden erfasst und in ein digitales Sprachsignal gewandelt wird, wobei das digitale Sprachsignal in seinem zeitlichen Verlauf analysiert wird, um
a. In dem zeitlichen Verlauf des digitalen Sprachsignals einen Zeitpunkt des Stimmeinsatzes des Sprechenden zu bestimmen,
b. eine Grundfrequenz des Sprachsignals zum Zeitpunkt des Stimmeinsatzes zu ermitteln,
c. aus dem digitalen Sprachsignal in einem vorgegebenen Zeitintervall ab dem Zeitpunkt des Stimmeinsatzes zeitlich aufgelöst den Verlauf der bei der Grundfrequenz in dem Sprachsignal enthaltenen Energie zu ermitteln;
d. aus dem digitalen Sprachsignal in dem vorgegebenen Zeitintervall zeitlich aufgelöst den Verlauf der bei wenigstens einem harmonischen Vielfachen der Grundfrequenz in dem Sprachsignal enthaltenen Energie zu ermitteln;
e. den zeitlichen Verlauf des Verhältnisses der in Schritten c und d ermittelten Energien zu bestimmen,
wobei auf einen weichen Stimmeinsatz geschlossen wird, wenn in dem Zeitintervall das gemäß vorstehenden Schritt e. ermittelte Verhältnis der Energien zunächst von der Energie bei der Grundfrequenz dominiert wird, erst im weiteren Verlauf des vorgegebenen Zeitintervalls in einer Zeitspanne Δt das Verhältnis der Energien sich zugunsten der Energie(n) bei der/den harmonischen Vielfachen der Grundfrequenz verschiebt.
Weiterhin offenbart werden ein Datenverarbeitungsprogramm für die rechnergestützte, automatisierte Beurteilung einer Qualität eines Stimmeinsatzes eines Sprechenden, das ein solches Verfahren umsetzt, sowie ein rechnergestütztes Datenverarbeitungsgerät mit einem solchen Datenverarbeitungsprogramm.
Mit der Erfindung lässt sich die Qualität eines Stimmeinsatzes eines Sprechenden sehr viel besser und genauer automatisiert beurteilen, als dies nach dem Stand der Technik möglich war.
摘要翻译：公开的是判断语音使用的扬声器，其特征在于，被检测并转换成数字语音信号的扬声器的声学语音信号的质量的方法，所述数字语音信号在它的时间特性的分析。在数字语音信号的时间过程中确定讲话者的语音使用时间，b。确定语音应用时语音信号的基频，c。从语音以时间分辨形式使用的时间起在预定时间间隔内从数字语音信号中确定包含在语音信号中的基频中的能量的进程; d。从在预定的时间间隔，以确定包含在语音信号中的基本频率中的至少一个谐波倍数的能量的时间分辨文件，与数字语音信号; 即确定在步骤c和d中确定的能量比率的时间进程，其中如果在该时间间隔内进行上述步骤e，则得出使用软语音的结论。能量的确定的比率最初由能量在基频为主，只在预定的时间间隔的剩余部分在一段时间.DELTA.t的能量的有利于在基本频率偏移的/谐波倍数的能量（或多个）的比率。进一步公开了一种用于使用语音的扬声器，它转换这样的方法，和一种基于计算机的数据处理装置与这样的数据处理程序中的质量的计算机辅助，自动评估的数据处理程序。通过本发明，与现有技术相比，可以评估讲话者语音使用的质量好得多且更精确的自动化。

19. 发明公开

EP3161825A1 TEMPORAL GAIN ADJUSTMENT BASED ON HIGH-BAND SIGNAL CHARACTERISTIC 有权
公开(公告)号：EP3161825A1
公开(公告)日：2017-05-03
申请号：EP15731780.1
申请日：2015-06-05
申请人： Qualcomm Incorporated
发明人： ATTI, Venkatraman S. , KRISHNAN, Venkatesh , RAJENDRAN, Vivek , CHEBIYYAM, Venkata Subrahmanyam Chandra Sekhar , SUBASINGHA, Subasingha Shaminda
IPC分类号： G10L21/038 , G10L19/24 , G10L19/02 , G10L19/06
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.

20. 发明公开

EP2954513A1 AMBIENT NOISE ROOT MEAN SQUARE (RMS) DETECTOR 有权
标题翻译：探测器方形环境噪声的方式
公开(公告)号：EP2954513A1
公开(公告)日：2015-12-16
申请号：EP13783126.9
申请日：2013-07-04
申请人： Cirrus Logic, Inc.
发明人： MILANI, Ali, Abdollahzadeh
IPC分类号： G10K11/178 , G10L21/0224
CPC分类号： H04R29/00 , G10K11/178 , G10K2210/108 , G10K2210/3023 , G10L21/0216 , G10L21/0224 , G10L2021/02165 , H04R29/004
摘要： An RMS detector uses the concept of the k-NN (classifying using nearest neighbors)—algorithm in order to obtain RMS values. A rms detector using first-order regressor with a variable smoothing factor is modified to penalize samples from center of data in order to obtain RMS values. Samples which vary greatly from the background noise levels, such as speech, scratch, wind and other noise spikes, are dampened in the RMS calculation. When background noise changes, the system will track the changes in background noise and include the changes in the calculation of the corrected RMS value. A minimum tracker runs more often (e.g. two or three times) than the rate as in prior art detectors and methods, tracks the minimum rms value, which is to compute a normalized distance value, which in turn is used to normalize the smoothing factor. From this data, a corrected or revised RMS value is determined as the function of the previous RMS value multiplied by one minus the smoothing factor plus the smooth factor times the minimum rms value to output the corrected RMS for the present invention. The rms value is used to generate a reset signal for the minimum tracker and is used to avoid deadlock in the tracker, for example, when the background signal increases/decreases over time.

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