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公开(公告)号：EP2672684B1
公开(公告)日：2018-10-17
申请号：EP12830214.8
申请日：2012-09-06
发明人： ZHANG, Jinchang
IPC分类号： H04M1/725 , H04M1/60 , G10L21/00 , G10L21/0364 , G10L21/034 , G10L25/03
CPC分类号： H04W88/06 , G10L21/00 , G10L21/034 , G10L21/0364 , G10L25/03
摘要： Embodiments of the present invention disclose a call method of a terminal and a terminal using the call method, relate to the field of communications technologies, and are invented for adjusting voice quality of a call in real time. The method includes: analyzing a spectral component of a voice signal during a call; and selecting a corresponding frequency response channel according to an analysis result of the spectral component of the voice signal. The present invention is mainly applied in the field of call services.
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3.发明公开
公开(公告)号：EP3350980A1
公开(公告)日：2018-07-25
申请号：EP16847147.2
申请日：2016-09-13
申请人： Cogito Corporation
发明人： FEAST, Joshua , AZARBAYEJANI, Ali , PLACE, Skyler
IPC分类号： H04M3/56
CPC分类号： H04M3/2281 , G10L25/03 , G10L25/72 , H04M3/2218 , H04M3/2227 , H04M3/562 , H04M7/0069 , H04M13/00 , H04M2203/355 , H04M2250/16
摘要： Systems and methods are provided for managing and analyzing multi-party dialogs (e.g., call) between communication devices. A digital connection is established with each of a plurality of communication devices. The connection is switched between the communication devices from a POTS connection to digital connections, enabling the communication devices to communicate with each other via the computing device over the digital connections. Audio signals are part of a multi-party dialog between users of the plurality of communication devices. The received audio signals are split into corresponding first signals and second signals. The first signals are transmitted to the plurality of communication devices and are analyzed to produce measurements of features of the second signals. Feedback data is transmitted to at least one of the plurality of communication devices.
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公开(公告)号：EP3252768A4
公开(公告)日：2018-06-27
申请号：EP16743427
申请日：2016-01-27
摘要： A parameter determination device comprises: a spectral envelope estimating portion 42 performing estimation of a spectral envelope using a parameter · 0 specified in a predetermined method, regarding the · 0 -th power of absolute values of a frequency domain sample sequence corresponding to a time-series signal as a power spectrum on the assumption that the parameter · 0 and a parameter · are positive numbers; a whitened spectral sequence generating portion 43 obtaining a whitened spectral sequence which is a sequence obtained by dividing the frequency domain sample sequence by the spectral envelope; and a parameter acquiring portion 44 determining such a parameter · that generalized Gaussian distribution with the parameter · as a shape parameter approximates a histogram of the whitened spectral sequence.
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公开(公告)号：EP3309777A1
公开(公告)日：2018-04-18
申请号：EP16306350.6
申请日：2016-10-13
申请人： Thomson Licensing
发明人： GILBERTON, Philippe , KITIC, Srdan
CPC分类号： G10L15/02 , G06F17/30743 , G10L19/02 , G10L25/03 , G10L25/18 , G10L25/45 , G10L25/93 , G10L2025/937
摘要： A device (200) and method for calculating scattering features for audio signal recognition. An interface (240) receives an audio signal that is processed (S610) by a processor (210) to obtain an audio frame. The processor (210) calculates (S620) a first order scattering features from at least one audio frame and then calculates (S630) for the first order scattering features an estimation of whether the first order scattering features comprises sufficient information for accurate audio signal recognition. The processor (240) calculates (S650) a second order scattering features from the first order scattering features only in case the first order scattering features does not comprise sufficient information for accurate audio signal recognition. As second order features are calculated only when it is deemed necessary, less processing power can be used by the device, which can lead to less power used by the device.
摘要翻译： 一种用于计算音频信号识别的散射特征的设备（200）和方法。 接口（240）接收由处理器（210）处理（S610）的音频信号以获得音频帧。 处理器（210）从至少一个音频帧计算（S620）一阶散射特征，然后针对一阶散射特征计算（S630）对一阶散射特征是否包括用于精确音频信号识别的足够信息的估计。 仅当第一阶散射特征不包含用于精确音频信号识别的足够信息时，处理器（240）才从第一阶散射特征计算（S650）第二阶散射特征。 由于二阶特征仅在必要时才被计算，所以设备可以使用较少的处理能力，这可以导致设备使用较少的功率。
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公开(公告)号：EP3288022A1
公开(公告)日：2018-02-28
申请号：EP17195678.2
申请日：2013-12-18
IPC分类号： G10H1/36 , G10H1/38 , G10L25/90 , G10L21/013 , G10L25/03
CPC分类号： G10L21/013 , G10H1/366 , G10H1/383 , G10H2210/021 , G10H2210/331 , G10L25/03 , G10L25/90 , G10L2025/906
摘要： Systems and methods for adjusting pitch of an audio signal include detecting input notes in the audio signal, mapping the input notes to corresponding output notes, each output note having an associated upper note boundary and lower note boundary, and modifying at least one of the upper note boundary and the lower note boundary of at least one output note in response to previously received input notes. Pitch of the input notes may be shifted to match an associated pitch of corresponding output notes. Delay of the pitch shifting process may be dynamically adjusted based on detected stability of the input notes.
摘要翻译： 用于调节音频信号的音高的系统和方法包括检测音频信号中的输入音符，将输入音符映射到对应的输出音符，每个输出音符具有相关联的上音调边界和下音调边界，并且修改上音调边界中的至少一个 音符边界和至少一个输出音符的低音符边界，以响应先前接收到的输入音符。 输入音符的音高可以被移动以匹配相应输出音符的相关音高。 可以基于检测到的输入音符的稳定性来动态调整音高移位过程的延迟。
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公开(公告)号：EP3144933A4
公开(公告)日：2017-03-22
申请号：EP15811228
申请日：2015-06-23
申请人： HUAWEI TECH CO LTD
发明人： WANG ZHE
CPC分类号： G10L19/22 , G10L19/02 , G10L19/0204 , G10L19/035 , G10L19/04 , G10L19/06 , G10L19/20 , G10L25/03
摘要： An audio encoding method and an apparatus are provided. The method includes: determining sparseness of distribution, on spectrums, of energy of N input audio frames (101), where the N audio frames include a current audio frame, and N is a positive integer; and determining, according to the sparseness of distribution, on the spectrums, of the energy of the N audio frames, whether to use a first encoding method or a second encoding method to encode the current audio frame (102), where the first encoding method is an encoding method that is based on time-frequency transform and transform coefficient quantization and that is not based on linear prediction, and the second encoding method is a linear-predication-based encoding method. According to the method, when an audio frame is encoded, sparseness of distribution, on a spectrum, of energy of the audio frame is considered, which can reduce encoding complexity and ensure that encoding is of relatively high accuracy.
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公开(公告)号：EP3105756A1
公开(公告)日：2016-12-21
申请号：EP15749174.7
申请日：2015-02-13
CPC分类号： G10L21/06 , G06F3/016 , G10L21/0264 , G10L21/0364 , G10L25/03 , G10L25/78 , G10L25/93 , G10L2025/783
摘要： An audio perception system is described, comprising a capture module configured to capture acoustic speech signal information; a feature extraction module configured to extract features that identify a candidate unvoiced portion in an acoustic signal; a classification module configured to identify if the acoustic signal is or contains an unvoiced portion based on the extracted features; and a control module configured to generate a control signal to a sensory stimulation actuator for generating an aero-tactile stimulation to be delivered to a listener, the control signal based at least in part on a signal representing the identified unvoiced portion. Related methods are also described.
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9.发明公开METHODS OF DECODING SPEECH FROM THE BRAIN AND SYSTEMS FOR PRACTICING THE SAME 审中-公开方法话音编码申请大脑和系统THEREOF
公开(公告)号：EP2959280A4
公开(公告)日：2016-10-19
申请号：EP14754973
申请日：2014-02-19
CPC分类号： G10L21/00 , A61B5/04001 , A61B5/4076 , A61B5/4094 , A61B5/4803 , A61B5/7203 , A61B5/7257 , A61B5/726 , G09B21/00 , G10L13/043 , G10L15/24 , G10L25/03 , G10L25/06
摘要： Provided are methods of decoding speech from the brain of a subject. The methods include detecting speech production signals from electrodes operably coupled to the speech motor cortex of a subject while the subject produces or imagines producing a speech sound. The methods further include deriving a speech production signal pattern from the detected speech production signals, and correlating the speech production signal pattern with a reference speech production signal pattern to decode speech from the brain of the subject. Speech communication systems and devices for practicing the subject methods are also provided.
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10.发明公开ADAPTIVE SPEECH FILTER FOR ATTENUATION OF AMBIENT NOISE 审中-公开ADAPTIVER SPRACHFILTER ZURUNTERDRÜCKUNGVONUMGEBUNGSGERÄUSCHEN
公开(公告)号：EP3032536A1
公开(公告)日：2016-06-15
申请号：EP15198584.3
申请日：2015-12-09
发明人： Herberger, Tilman , Tost, Titus
IPC分类号： G10L21/0264 , G10L25/03
CPC分类号： G10L21/0264 , G10L25/03
摘要： According to a preferred aspect of the instant invention, there is provided a system and method that allows the user to attenuate ambient noise in speech recordings in the audio part of a video recording. The user does not need to define particular sections or samples or individual parameters. The system is automatically analyzing the input signal and in a plurality of individual steps detects the ambient noise, determines an adaptive filter, implements the filter and therewith attenuates the ambient noise accordingly.
摘要翻译： 根据本发明的优选方面，提供了一种系统和方法，其允许用户在视频记录的音频部分中的语音记录中衰减环境噪声。 用户不需要定义特定的部分或样本或各个参数。 该系统自动分析输入信号，并且在多个单独步骤中检测环境噪声，确定自适应滤波器，实现滤波器并因此相应地衰减环境噪声。
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