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1. 发明申请

WO2019149845A1 AUDIO SCENE ENCODER, AUDIO SCENE DECODER AND RELATED METHODS USING HYBRID ENCODER/DECODER SPATIAL ANALYSIS 审中-公开
公开(公告)号：WO2019149845A1
公开(公告)日：2019-08-08
申请号：PCT/EP2019/052428
申请日：2019-01-31
申请人： FRAUNHOFER-GESELLSCHAFT ZUR FÖRDERUNG DER ANGEWANDTEN FORSCHUNG E.V. , FRIEDRICH-ALEXANDER-UNIVERSITÄT ERLANGEN-NÜRNBERG
发明人： FUCHS, Guillaume , BAYER, Stefan , MULTRUS, Markus , THIERGART, Oliver , BOUTHÉON, Alexandre , HERRE, Jürgen , GHIDO, Florin , JAEGERS, Wolfgang , KÜCH, Fabian
IPC分类号： G10L19/008 , G10L19/18 , G10L19/02
摘要： An audio scene encoder for encoding an audio scene, the audio scene comprising at least two component signals, comprises: a core encoder (160) for core encoding the at least two component signals, wherein the core encoder (160) is configured to generate a first encoded representation (310) for a first portion of the at least two component signals, and to generate a second encoded representation (320) for a second portion of the at least two component signals, a spatial analyzer (200) for analyzing the audio scene to derive one or more spatial parameters (330) or one or more spatial parameter sets for the second portion; and an output interface (300) for forming the encoded audio scene signal (340), the encoded audio scene signal (340) comprising the first encoded representation (310), the second encoded representation (320), and the one or more spatial parameters (330) or one or more spatial parameter sets for the second portion.

2. 发明申请

WO2019097017A1 APPARATUS AND METHOD FOR ENCODING OR DECODING DIRECTIONAL AUDIO CODING PARAMETERS USING DIFFERENT TIME/FREQUENCY RESOLUTIONS 审中-公开
公开(公告)号：WO2019097017A1
公开(公告)日：2019-05-23
申请号：PCT/EP2018/081620
申请日：2018-11-16
申请人： FRAUNHOFER-GESELLSCHAFT ZUR FÖRDERUNG DER ANGEWANDTEN FORSCHUNG E.V. , FRIEDRICH-ALEXANDER-UNIVERSITÄT ERLANGEN-NÜRNBERG
发明人： FUCHS, Guillaume , HERRE, Jürgen , KÜCH, Fabian , DÖHLA, Stefan , MULTRUS, Markus , THIERGART, Oliver , WÜBBOLT, Oliver , GHIDO, Florin , BAYER, Stefan , JÄGERS, Wolfgang
IPC分类号： G10L19/00 , G10L19/008
摘要： An apparatus for encoding directional audio coding parameters comprising diffuseness parameters and direction parameters, comprises: a parameter calculator (100) for calculating the diffuseness parameters with a first time or frequency resolution and for calculating the direction parameters with a second time or frequency resolution; and a quantizer and encoder processor (200) for generating a quantized and encoded representation of the diffuseness parameters and the direction parameters.

3. 发明申请

WO2012072798A1 SOUND ACQUISITION VIA THE EXTRACTION OF GEOMETRICAL INFORMATION FROM DIRECTION OF ARRIVAL ESTIMATES 审中-公开
标题翻译：通过从到达估计方向提取几何信息的声音收购
公开(公告)号：WO2012072798A1
公开(公告)日：2012-06-07
申请号：PCT/EP2011/071629
申请日：2011-12-02
申请人： FRAUNHOFER-GESELLSCHAFT ZUR FÖRDERUNG DER ANGEWANDTEN FORSCHUNG E.V. , FRIEDRICH-ALEXANDER-UNIVERSITÄT ERLANGEN-NÜRNBERG , HERRE, Jürgen , KÜCH, Fabian , KALLINGER, Markus , DEL GALDO, Giovanni , THIERGART, Oliver , MAHNE, Dirk , KUNTZ, Achim , KRATSCHMER, Michael , CRACIUN, Alexandra
发明人： HERRE, Jürgen , KÜCH, Fabian , KALLINGER, Markus , DEL GALDO, Giovanni , THIERGART, Oliver , MAHNE, Dirk , KUNTZ, Achim , KRATSCHMER, Michael , CRACIUN, Alexandra
IPC分类号： H04R3/00 , G10L19/00
CPC分类号： G10L19/00 , G10L19/008 , G10L19/02 , G10L19/167 , G10L19/20 , H04R1/326 , H04R3/005 , H04R2430/21
摘要： An apparatus for generating an audio output signal to simulate a recording of a virtual microphone at a configurable virtual position in an environment is provided. The apparatus comprises a sound events position estimator and an information computation module (120). The sound events position estimator (110) is adapted to estimate a sound source position indicating a position of a sound source in the environment, wherein the sound events position estimator (110) is adapted to estimate the sound source position based on a first direction information provided by a first real spatial microphone being located at a first real microphone position in the environment, and based on a second direction information provided by a second real spatial microphone being located at a second real microphone position in the environment. The information computation module (120) is adapted to generate the audio output signal based on a first recorded audio input signal, based on the first real microphone position, based on the virtual position of the virtual microphone, and based on the sound source position.
摘要翻译：提供了一种用于生成音频输出信号以模拟在环境中的可配置虚拟位置处的虚拟麦克风的记录的装置。该装置包括声音事件位置估计器和信息计算模块（120）。声音事件位置估计器（110）适于估计指示环境中声源的位置的声源位置，其中声音事件位置估计器（110）适于基于第一方向信息来估计声源位置由位于环境中的第一实际麦克风位置处的第一实际空间麦克风提供，并且基于由位于环境中的第二真实麦克风位置的第二实际空间麦克风提供的第二方向信息。信息计算模块（120）适于基于第一真实麦克风位置，基于虚拟麦克风的虚拟位置，并且基于声源位置，基于第一记录音频输入信号生成音频输出信号。

4. 发明申请

WO2014095250A1 FILTER AND METHOD FOR INFORMED SPATIAL FILTERING USING MULTIPLE INSTANTANEOUS DIRECTION-OF-ARRIVIAL ESTIMATES 审中-公开
标题翻译：使用多个即时方向估计的通知空间滤波的滤波器和方法
公开(公告)号：WO2014095250A1
公开(公告)日：2014-06-26
申请号：PCT/EP2013/074650
申请日：2013-11-25
申请人： FRAUNHOFER-GESELLSCHAFT ZUR FÖRDERUNG DER ANGEWANDTEN FORSCHUNG E.V. , FRIEDRICH-ALEXANDER-UNIVERSITÄT ERLANGEN-NÜRNBERG
发明人： HABETS, Emanuel , THIERGART, Oliver , BRAUN, Sebastian , TASESKA, Maja
IPC分类号： H04R3/00 , G10K11/34
CPC分类号： G06F3/16 , G10K11/346 , H04R3/005 , H04R3/02
摘要： A filter (100) for generating an audio output signal, comprising a plurality of audio output signal samples, based on two or more input microphone signals is provided. The audio output signal and the two or more input microphone signals are represented in a time-frequency domain, wherein each of the plurality of audio output signal samples is assigned to a time-frequency bin ((k, n)) of a plurality of time-frequency bins ((k, n)). The filter (100) comprises a weights generator (110) being adapted to receive, for each of the plurality of time-frequency bins ((k, n)), direction-of-arrival information of one or more sound components of one or more sound sources or position information of one or more sound sources, and being adapted to generate weighting information for each of the plurality of time-frequency bins ((k, n)) depending on the direction-of-arrival information of the one or more sound components of the one more sound sources of said time- frequency bin ((k, n)) or depending on the position information of the one or more sound sources of said time-frequency bin ((k n)). Moreover, the filter comprises an output signal generator (120) for generating the audio output signal by generating for each of the plurality of time-frequency bins ((k, n)) one of the plurality of audio output signal samples, which is assigned to said time-frequency bin ((k, n)), depending on the weighting information of said time-frequency bin ((k, n)) and depending on an audio input sample, being assigned to said time-frequency bin ((k, n)), of each of the two or more input microphone signals.
摘要翻译：提供了一种用于基于两个或多个输入麦克风信号产生包括多个音频输出信号样本的音频输出信号的滤波器（100）。音频输出信号和两个或更多个输入麦克风信号以时间 - 频率域表示，其中多个音频输出信号样本中的每一个被分配给多个音频输出信号样本的时间频率分组（（k，n））时频箱（（k，n））。滤波器（100）包括权重发生器（110），其适于为多个时频频段（（k，n））中的每一个接收一个或多个时频频段的一个或多个声音分量的到达方向信息，更多的声源或一个或多个声源的位置信息，并且适于根据所述一个或多个声源的到达方向信息而生成用于多个时频频带（（k，n））中的每一个的加权信息所述时频仓（（k，n））的另一个声源的更多声音分量或者根据所述时间频率仓（（kn））的一个或多个声源的位置信息。此外，滤波器包括输出信号发生器（120），用于通过为多个音频输出信号样本中的一个音频输出信号样本（（k，n））中的每一个生成音频输出信号，根据所述时间频率bin（（k，n））的加权信息，并且根据音频输入样本，分配给所述时间频率bin（（k，n））， k，n）），两个或更多个输入麦克风信号中的每一个。

5. 发明申请

WO2019097018A1 APPARATUS AND METHOD FOR ENCODING OR DECODING DIRECTIONAL AUDIO CODING PARAMETERS USING QUANTIZATION AND ENTROPY CODING 审中-公开
公开(公告)号：WO2019097018A1
公开(公告)日：2019-05-23
申请号：PCT/EP2018/081623
申请日：2018-11-16
申请人： FRAUNHOFER-GESELLSCHAFT ZUR FÖRDERUNG DER ANGEWANDTEN FORSCHUNG E.V. , FRIEDRICH-ALEXANDER-UNIVERSITÄT ERLANGEN-NÜRNBERG
发明人： FUCHS, Guillaume , HERRE, Jürgen , KÜCH, Fabian , DÖHLA, Stefan , MULTRUS, Markus , THIERGART, Oliver , WÜBBOLT, Oliver , GHIDO, Florin , BAYER, Stefan , JÄGERS, Wolfgang
IPC分类号： G10L19/00 , G10L19/008
摘要： An apparatus for encoding directional audio coding parameters comprising diffuseness parameters and direction parameters, comprises: a parameter quantizer (210) for quantizing the diffuseness parameters and the direction parameters; a parameter encoder (220) for encoding quantized diffuseness parameters and quantized direction parameters; and an output interface (230) for generating an encoded parameter representation comprising information on encoded diffuseness parameters and encoded direction parameters.

6. 发明申请

WO2015169617A1 SYSTEM, APPARATUS AND METHOD FOR CONSISTENT ACOUSTIC SCENE REPRODUCTION BASED ON ADAPTIVE FUNCTIONS 审中-公开
标题翻译：基于自适应函数的一致性声场再现的系统，装置和方法
公开(公告)号：WO2015169617A1
公开(公告)日：2015-11-12
申请号：PCT/EP2015/058857
申请日：2015-04-23
申请人： FRAUNHOFER-GESELLSCHAFT ZUR FÖRDERUNG DER ANGEWANDTEN FORSCHUNG E.V. , FRIEDRICH-ALEXANDER-UNIVERSITÄT ERLANGEN-NÜRNBERG
发明人： HABETS, Emanuel , THIERGART, Oliver , KOWALCZYK, Konrad
IPC分类号： H04S7/00 , H04R25/00
CPC分类号： H04S5/005 , G10L19/008 , H04R25/407 , H04R25/552 , H04S7/30 , H04S7/303 , H04S7/307 , H04S2400/11 , H04S2400/13 , H04S2400/15
摘要： A system for generating one or more audio output signals is provided. The system comprises a decomposition module (101), a signal processor (105), and an output interface (106). The signal processor (105) is configured to receive the direct component signal, the diffuse component signal and direction information, said direction information depending on a direction of arrival of the direct signal components of the two or more audio input signals. Moreover, the signal processor (105) is configured to generate one or more processed diffuse signals depending on the diffuse component signal. For each audio output signal of the one or more audio output signals, the signal processor (105) is configured to determine, depending on the direction of arrival, a direct gain, the signal processor (105) is configured to apply said direct gain on the direct component signal to obtain a processed direct signal, and the signal processor (105) is configured to combine said processed direct signal and one of the one or more processed diffuse signals to generate said audio output signal. The output interface (106) is configured to output the one or more audio output signals. The signal processor (105) comprises a gain function computation module (104) for calculating one or more gain functions, wherein each gain function of the one or more gain functions, comprises a plurality of gain function argument values, wherein a gain function return value is assigned to each of said gain function argument values, wherein, when said gain function receives one of said gain function argument values, wherein said gain function is configured to return the gain function return value being assigned to said one of said gain function argument values. Moreover, the signal processor (105) further comprises a signal modifier (103) for selecting, depending on the direction of arrival, a direction dependent argument value from the gain function argument values of a gain function of the one or more gain functions, for obtaining the gain function return value being assigned to said direction dependent argument value from said gain function, and for determining the gain value of at least one of the one or more audio output signals depending on said gain function return value obtained from said gain function.
摘要翻译：提供了一种用于产生一个或多个音频输出信号的系统。该系统包括分解模块（101），信号处理器（105）和输出接口（106）。信号处理器（105）被配置为根据两个或多个音频输入信号的直接信号分量的到达方向接收直接分量信号，扩散分量信号和方向信息，所述方向信息。此外，信号处理器（105）被配置为根据扩散分量信号产生一个或多个经处理的扩散信号。对于一个或多个音频输出信号的每个音频输出信号，信号处理器（105）被配置为根据到达方向确定直接增益，信号处理器（105）被配置为将所述直接增益应用于所述直接分量信号以获得经处理的直接信号，并且所述信号处理器（105）被配置为将所述经处理的直接信号与所述一个或多个经处理的扩散信号中的一个组合以产生所述音频输出信号。输出接口（106）被配置为输出一个或多个音频输出信号。信号处理器（105）包括用于计算一个或多个增益函数的增益函数计算模块（104），其中所述一个或多个增益函数的每个增益函数包括多个增益函数自变量值，其中增益函数返回值被分配给每个所述增益函数自变量值，其中当所述增益函数接收到所述增益函数自变量值之一时，其中所述增益函数被配置为返回分配给所述增益函数自变量值中的所述一个增益函数参数值的增益函数返回值。此外，信号处理器（105）还包括信号修改器（103），用于根据到达方向，根据一个或多个增益函数的增益函数的增益函数自变量值来选择与方向相关的参数值，用于从所述增益函数获得分配给所述方向相关参数值的增益函数返回值，并且根据从所述增益函数获得的所述增益函数返回值来确定所述一个或多个音频输出信号中的至少一个的增益值。

7. 发明申请

WO2012072804A1 APPARATUS AND METHOD FOR GEOMETRY-BASED SPATIAL AUDIO CODING 审中-公开
标题翻译：用于基于几何的空间音频编码的装置和方法
公开(公告)号：WO2012072804A1
公开(公告)日：2012-06-07
申请号：PCT/EP2011/071644
申请日：2011-12-02
申请人： Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. , Friedrich-Alexander-Universität Erlangen-Nürnberg , DEL GALDO, Giovanni , THIERGART, Oliver , HERRE, Jürgen , KÜCH, Fabian , HABETS, Emanuel , CRACIUN, Alexandra , KUNTZ, Achim
发明人： DEL GALDO, Giovanni , THIERGART, Oliver , HERRE, Jürgen , KÜCH, Fabian , HABETS, Emanuel , CRACIUN, Alexandra , KUNTZ, Achim
IPC分类号： G10L19/14 , G10L19/02 , G10L19/00 , H04R3/00
CPC分类号： G10L19/00 , G10L19/008 , G10L19/02 , G10L19/167 , G10L19/20 , H04R1/326 , H04R3/005 , H04R2430/21
摘要： An apparatus for generating at least one audio output signal based on an audio data stream comprising audio data relating to one or more sound sources is provided. The apparatus comprises a receiver for receiving the audio data stream comprising the audio data. The audio data comprises one or more pressure values for each one of the sound sources. Furthermore, the audio data comprises one or more position values indicating a position of one of the sound sources for each one of the sound sources. Moreover, the apparatus comprises a synthesis module for generating the at least one audio output signal based on at least one of the one or more pressure values of the audio data of the audio data stream and based on at least one of the one or more position values of the audio data of the audio data stream.
摘要翻译：提供了一种用于基于包括与一个或多个声源相关的音频数据的音频数据流来生成至少一个音频输出信号的装置。该装置包括用于接收包括音频数据的音频数据流的接收器。音频数据包括每个声源的一个或多个压力值。此外，音频数据包括指示每个声源的声源中的一个的位置的一个或多个位置值。此外，该装置包括合成模块，用于基于音频数据流的音频数据的一个或多个压力值中的至少一个产生至少一个音频输出信号，并且基于该一个或多个位置音频数据流的音频数据的值。

8. 发明申请

WO2017157803A1 APPARATUS, METHOD OR COMPUTER PROGRAM FOR GENERATING A SOUND FIELD DESCRIPTION 审中-公开
标题翻译：用于产生声场的装置，方法或计算机程序
公开(公告)号：WO2017157803A1
公开(公告)日：2017-09-21
申请号：PCT/EP2017/055719
申请日：2017-03-10
申请人： FRAUNHOFER-GESELLSCHAFT ZUR FÖRDERUNG DER ANGEWANDTEN FORSCHUNG E.V. , FRIEDRICH-ALEXANDER-UNIVERSITÄT ERLANGEN-NÜRNBERG
发明人： HABETS, Emanuel , THIERGART, Oliver , KÜCH, Fabian , NIEDERLEITNER, Alexander , KHAN, Affan-Hasan , MAHNE, Dirk
IPC分类号： H04R3/00
摘要： An apparatus for generating a sound field description having a representation of sound field components, comprises a direction determiner (102) for determining one or more sound directions for each time-frequency tile of a plurality of time-frequency tiles of a plurality of microphone signals; a spatial basis function evaluator (103) for evaluating, for each time-frequency tile of the plurality of time-frequency tiles, one or more spatial basis functions using the one or more sound directions; and a sound field component calculator (201 ) for calculating, for each time-frequency tile of the plurality of time-frequency tiles, one or more sound field components corresponding to the one or more spatial basis functions evaluated using the one or more sound directions and a reference signal for a corresponding time-frequency tile, the reference signal being derived from one or more microphone signals of the plurality of microphone signals.
摘要翻译：用于产生具有声场分量的表示的声场描述的设备包括方向确定器（102），用于确定多个时间 - 频率块中的每个时间 - 频率块的一个或多个声音方向，多个麦克风信号的频率块; 空间基础函数评估器（103），用于针对所述多个时间频率片中的每个时间频率片评估使用所述一个或多个声音方向的一个或多个空间基础函数; 以及声场分量计算器（201），用于针对所述多个时频分片中的每个时频分片，计算与使用所述一个或多个声音方向评估的所述一个或多个空间基函数相对应的一个或多个声场分量以及用于对应的时间频率片的参考信号，所述参考信号是从所述多个麦克风信号中的一个或多个麦克风信号中导出的。

9. 发明申请

WO2015169618A1 SYSTEM, APPARATUS AND METHOD FOR CONSISTENT ACOUSTIC SCENE REPRODUCTION BASED ON INFORMED SPATIAL FILTERING 审中-公开
标题翻译：基于信息空间滤波的一致性声场再现系统，装置和方法
公开(公告)号：WO2015169618A1
公开(公告)日：2015-11-12
申请号：PCT/EP2015/058859
申请日：2015-04-23
申请人： FRAUNHOFER-GESELLSCHAFT ZUR FÖRDERUNG DER ANGEWANDTEN FORSCHUNG E.V. , FRIEDRICH-ALEXANDER-UNIVERSITÄT ERLANGEN-NÜRNBERG
发明人： HABETS, Emanuel , THIERGART, Oliver , KOWALCZYK, Konrad
IPC分类号： H04S7/00 , H04R25/00
CPC分类号： H04S5/005 , G10L19/008 , H04R25/407 , H04R25/552 , H04S7/30 , H04S7/303 , H04S7/307 , H04S2400/11 , H04S2400/13 , H04S2400/15
摘要： A system for generating one or more audio output signals is provided. The system comprises a decomposition module (101), a signal processor (105), and an output interface (106). The decomposition module (101) is configured to receive two or more audio input signals, wherein the decomposition module (101) is configured to generate a direct component signal, comprising direct signal components of the two or more audio input signals, and wherein the decomposition module (101) is configured to generate a diffuse component signal, comprising diffuse signal components of the two or more audio input signals. The signal processor (105) is configured to receive the direct component signal, the diffuse component signal and direction information, said direction information depending on a direction of arrival of the direct signal components of the two or more audio input signals. Moreover, the signal processor (105) is configured to generate one or more processed diffuse signals depending on the diffuse component signal. For each audio output signal of the one or more audio output signals, the signal processor (105) is configured to determine, depending on the direction of arrival, a direct gain, the signal processor (105) is configured to apply said direct gain on the direct component signal to obtain a processed direct signal, and the signal processor (105) is configured to combine said processed direct signal and one of the one or more processed diffuse signals to generate said audio output signal. The output interface (106) is configured to output the one or more audio output signals.
摘要翻译：提供了一种用于产生一个或多个音频输出信号的系统。该系统包括分解模块（101），信号处理器（105）和输出接口（106）。分解模块（101）被配置为接收两个或更多个音频输入信号，其中分解模块（101）被配置为生成直接分量信号，包括两个或多个音频输入信号的直接信号分量，并且其中分解模块（101）被配置为生成包括两个或更多个音频输入信号的漫射信号分量的漫射分量信号。信号处理器（105）被配置为根据两个或多个音频输入信号的直接信号分量的到达方向接收直接分量信号，扩散分量信号和方向信息，所述方向信息。此外，信号处理器（105）被配置为根据扩散分量信号产生一个或多个经处理的扩散信号。对于一个或多个音频输出信号的每个音频输出信号，信号处理器（105）被配置为根据到达方向确定直接增益，信号处理器（105）被配置为将所述直接增益应用于所述直接分量信号以获得经处理的直接信号，并且所述信号处理器（105）被配置为将所述经处理的直接信号与所述一个或多个经处理的扩散信号中的一个组合以产生所述音频输出信号。输出接口（106）被配置为输出一个或多个音频输出信号。

10. 发明申请

WO2015086377A1 EXTRACTION OF REVERBERANT SOUND USING MICROPHONE ARRAYS 审中-公开
标题翻译：使用麦克风阵列提取复音声音
公开(公告)号：WO2015086377A1
公开(公告)日：2015-06-18
申请号：PCT/EP2014/076252
申请日：2014-12-02
申请人： FRAUNHOFER-GESELLSCHAFT ZUR FÖRDERUNG DER ANGEWANDTEN FORSCHUNG E.V. , FRIEDRICH-ALEXANDER-UNIVERSITÄT ERLANGEN-NÜRNBERG
发明人： THIERGART, Oliver , HABETS, Emanuel
IPC分类号： G10L21/02 , G10L19/26 , H04R5/027 , G11B20/10
CPC分类号： G10L21/0264 , G10L19/26 , G10L21/02 , G10L21/0208 , G10L2021/02082 , G10L2021/02166 , G11B2020/10601 , H04R3/005 , H04R5/027 , H04S2400/15 , H04S2420/01
摘要： A method comprises estimating a spatial coherence between a first diffuse sound portion in a first microphone signal and a second diffuse sound portion in a second microphone signal. The first microphone signal is captured by a first microphone and the second microphone signal is captured by a second microphone wh ich is spaced apart from the first microphone in a known manner. The method further comprises defining a linear constraint for filter coefficients of a diffuse sound filter, the linear constraint being based on the spatial coherence. The method also comprises calculating at least one of signal statistics and noise statistics over the first microphone signal and the second microphone signal. The method also comprises determining the filter coefficients of the diffuse sound filter by solving an optimization problem concerning at least one of the signal statistics and noise statistics while considering the linear constraint for the filter coefficients.
摘要翻译：一种方法包括估计第一麦克风信号中的第一扩散声部分和第二麦克风信号中的第二扩散声部分之间的空间相干性。第一麦克风信号由第一麦克风捕获，并且第二麦克风信号由以已知方式与第一麦克风间隔开的第二麦克风捕获。该方法还包括定义扩散声音滤波器的滤波器系数的线性约束，线性约束基于空间相干性。该方法还包括通过第一麦克风信号和第二麦克风信号来计算信号统计和噪声统计中的至少一个。该方法还包括通过在考虑滤波器系数的线性约束的同时解决关于信号统计和噪声统计中的至少一个的优化问题来确定扩散声滤波器的滤波器系数。

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