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41. 发明申请

US20110211702A1 Signal Generation for Binaural Signals 有权
标题翻译：双耳信号的信号生成
公开(公告)号：US20110211702A1
公开(公告)日：2011-09-01
申请号：US13015335
申请日：2011-01-27
申请人： Harald Mundt , Bernhard Neugebauer , Johannes Hilpert , Andreas Silzle , Jan Plogsties
发明人： Harald Mundt , Bernhard Neugebauer , Johannes Hilpert , Andreas Silzle , Jan Plogsties
IPC分类号： H04R5/00
CPC分类号： H04S3/004 , H04S7/30 , H04S2400/01 , H04S2420/01
摘要： A device for generating a binaural signal based on a multi-channel signal representing a plurality of channels and intended for reproduction by a speaker configuration having a virtual sound source position associated to each channel, is described. It includes a correlation reducer for differently processing, and thereby reducing a correlation between, at least one of a left and a right channel of the plurality of channels, a front and a rear channel of the plurality of channels, and a center and a non-center channel of the plurality of channels, in order to obtain an inter-similarity reduced set of channels; a plurality of directional filters, a first mixer for mixing outputs of the directional filters modeling the acoustic transmission to the first ear canal of the listener, and a second mixer for mixing outputs of the directional filters modeling the acoustic transmission to the second ear canal of the listener. According to another aspect, a center level reduction for forming the downmix for a room processor is performed. According to even another aspect, an inter-similarity decreasing set of head-related transfer functions is formed.
摘要翻译：描述了一种用于基于表示多个信道并且由具有与每个信道相关联的虚拟声源位置的扬声器配置再现的多声道信号产生双耳信号的装置。它包括用于不同处理的相关减法器，从而减少多个通道中的左通道和右通道中的至少一个，多个通道的前通道和后通道之间的相关性，所述多个信道的中心信道，以便获得相互相似性减小的信道集合; 多个定向滤波器，用于混合将声学传播建模到收听者的第一耳道的方向滤波器的输出的第一混合器和用于混合将声学传播建模到第二耳道的方向滤波器的输出的第二混合器，听众。根据另一方面，执行用于形成室内处理器的下混合的中心水平降低。根据另一方面，形成头相关传递函数的相互相似性减小集合。

42. 发明申请

US20070129940A1 Method and apparatus for determining an estimate 有权
标题翻译：用于确定估计的方法和装置
公开(公告)号：US20070129940A1
公开(公告)日：2007-06-07
申请号：US11469418
申请日：2006-08-31
申请人： Michael Schug , Johannes Hilpert , Stefan Geyersberger , Max Neuendorf
发明人： Michael Schug , Johannes Hilpert , Stefan Geyersberger , Max Neuendorf
IPC分类号： G10L19/14
CPC分类号： G10L19/025 , G10L19/002
摘要： For determining an estimate of a need for information units for encoding a signal, a measure for the distribution of the energy in the frequency band is taken into account in addition to the admissible interference for a frequency band and an energy of the frequency band. With this, a better estimate of the need for information units is obtained, so that coding can be done more efficiently and more accurately.
摘要翻译：为了确定对信号进行编码的信息单元的需求的估计，除了对于频带的容许干扰和频带的能量之外，还考虑了频带中的能量分布的度量。因此，获得对信息单元的更好的估计，使得可以更有效和更准确地进行编码。

43. 发明申请

US20060235865A1 Entropy coding with compact codebooks 有权
标题翻译：使用紧凑型码本进行熵编码
公开(公告)号：US20060235865A1
公开(公告)日：2006-10-19
申请号：US11251485
申请日：2005-10-14
申请人： Ralph Sperschneider , Jurgen Herre , Karsten Linzmeier , Johannes Hilpert
发明人： Ralph Sperschneider , Jurgen Herre , Karsten Linzmeier , Johannes Hilpert
IPC分类号： G06F7/00
CPC分类号： H03M7/42
摘要： The present invention is based on the finding that an efficient code for encoding information values can be derived, when two or more information values are grouped in a tuple in a tuple order and when an encoding rule is used, that assigns the same code word to tuples having identical information values in different orders and that does derive an order information, indicating the tuple order, and when the code word is output in association with the order information.
摘要翻译：本发明基于以下发现：当两个或多个信息值以元组顺序被分组成元组并且当使用编码规则时，可以导出用于编码信息值的有效代码，将相同的代码字分配给元组具有不同顺序的相同信息值，并且确定导出指示元组顺序的订单信息，以及何时与订单信息相关联地输出代码字。

44. 发明申请

US20060235683A1 Lossless encoding of information with guaranteed maximum bitrate 审中-公开
标题翻译：无损编码具有保证最大比特率的信息
公开(公告)号：US20060235683A1
公开(公告)日：2006-10-19
申请号：US11233351
申请日：2005-09-22
申请人： Ralph Sperschneider , Jurgen Herre , Karsten Linzmeier , Johannes Hilpert
发明人： Ralph Sperschneider , Jurgen Herre , Karsten Linzmeier , Johannes Hilpert
IPC分类号： G10L19/02
CPC分类号： H03M7/30 , H03M7/3082
摘要： A compact encoded representation of information values not exceeding a predefined size can be derived when a first encoding rule generating an encoded representation of the information values of variable length is compared to a second encoding rule generating an encoded representation of the information values of fixed length and when the encoding rule resulting in the encoded representation requiring the lower number of information units is chosen. Thus, the maximum bit rate can be guaranteed to be at least the maximum bit rate of the second encoding rule deriving the second encoded representation. Signaling the choice of the encoding rule by some rule information together with the encoded representation of the information values, the correct information values can later on be derived on a decoder side, using a decoding rule fitting the encoding rule used during the encoding.
摘要翻译：当生成可变长度的信息值的编码表示的第一编码规则与产生固定长度的信息值的编码表示的第二编码规则进行比较时，可以导出不超过预定尺寸的信息值的紧凑编码表示，当选择导致需要较少数量的信息单元的编码表示的编码规则时。因此，可以保证最大比特率至少是导出第二编码表示的第二编码规则的最大比特率。通过一些规则信息与信息值的编码表示信号对编码规则的选择，使用适合编码期间使用的编码规则的解码规则，可以稍后在解码器侧导出正确的信息值。

45. 发明申请

US20060235679A1 Adaptive grouping of parameters for enhanced coding efficiency 有权
标题翻译：参数的自适应分组，以提高编码效率
公开(公告)号：US20060235679A1
公开(公告)日：2006-10-19
申请号：US11243915
申请日：2005-10-05
申请人： Ralph Sperschneider , Jurgen Herre , Karsten Linzmeier , Johannes Hilpert
发明人： Ralph Sperschneider , Jurgen Herre , Karsten Linzmeier , Johannes Hilpert
IPC分类号： G10L11/04
CPC分类号： H03M7/40 , G10L19/0017 , G10L19/002 , G10L19/008 , G10L19/0204 , G10L25/18 , H04N19/12 , H04N19/15 , H04N19/196 , H04N19/463 , H04N19/91
摘要： The present invention is based on the finding that parameters including a first set of parameters of a representation of a first portion of an original signal and including a second set of parameters of a representation of a second portion of the original signal can be efficiently encoded, when the parameters are arranged in a first sequence of tuples and in a second sequence of tuples, wherein the first sequence of tuples comprises tuples of parameters having two parameters from a single portion of the original signal and wherein the second sequence of tuples comprises tuples of parameters having one parameter from the first portion and one parameter from the second portion of the original signal. An efficient encoding can be achieved using a bit estimator to estimate the number of necessary bits to encode the first and the second sequence of tuples, wherein only the sequence of tuples is encoded, that results in the lower number of bits.
摘要翻译：本发明基于以下发现：可以有效地编码包括原始信号的第一部分的表示的第一组参数和包括原始信号的第二部分的表示的第二组参数的参数的发现，当所述参数被布置在元组的第一序列中并且以第二次元组排列时，其中所述元组的所述第一序列包括具有来自所述原始信号的单个部分的两个参数的参数的元组，并且其中所述第二序列元组包括：参数具有来自第一部分的一个参数和来自原始信号的第二部分的一个参数。可以使用位估计器来实现有效编码，以估计对第一和第二元组序列进行编码的必需位数，其中仅对元组的序列进行编码，导致较低位数。

46. 发明申请

US20050074127A1 Compatible multi-channel coding/decoding 有权
标题翻译：兼容多通道编码/解码
公开(公告)号：US20050074127A1
公开(公告)日：2005-04-07
申请号：US10679085
申请日：2003-10-02
申请人： Jurgen Herre , Johannes Hilpert , Stefan Geyersberger , Andreas Holzer , Claus Spenger
发明人： Jurgen Herre , Johannes Hilpert , Stefan Geyersberger , Andreas Holzer , Claus Spenger
IPC分类号： H04S3/00 , H04S3/02 , H04R5/00
CPC分类号： H04S3/02 , G10L19/008 , G10L19/032 , H04S3/008 , H04S2400/03 , H04S2420/03
摘要： In processing a multi-channel audio signal having at least three original channels, a first downmix channel and a second downmix channel are provided, which are derived from the original channels. For a selected original channel of the original channels, channel side information are calculated such that a downmix channel or a combined downmix channel including the first and the second downmix channels, when weighted using the channel side information, results in an approximation of the selected original channel. The channel side information and the first and second downmix channels form output data to be transmitted to a decoder, which, in case of a low level decoder only decodes the first and second downmix channels or, in case of a high level decoder provides a full multi-channel audio signal based on the downmix channels and the channel side information. Since the channel side information only occupy a low number of bits, and since the decoder does not use dematrixing, an efficient and high quality multi-channel extension for stereo players and enhanced multi-channel players is obtained.
摘要翻译：在处理具有至少三个原始信道的多声道音频信号时，提供从原始信道导出的第一下混通道和第二下混通道。对于原始频道的所选择的原始频道，计算频道侧信息，使得当使用频道侧信息加权时，包括第一和第二下混频道的下混频道或组合缩混频道导致所选原稿的近似渠道。信道侧信息以及第一和第二下混通道形成要发送到解码器的输出数据，其在低电平解码器的情况下仅解码第一和第二下混通道，或者在高电平解码器提供满基于下混频道的多声道音频信号和频道侧信息。由于信道侧信息仅占用少量的比特，并且由于解码器不使用解矩阵，因此可以获得用于立体声播放器和增强型多声道播放器的有效且高质量的多声道扩展。

47. 发明授权

US06826525B2 Method and device for detecting a transient in a discrete-time audio signal 有权
标题翻译：用于检测离散时间音频信号中的瞬变的方法和装置
公开(公告)号：US06826525B2
公开(公告)日：2004-11-30
申请号：US10183139
申请日：2002-06-25
申请人： Johannes Hilpert , Jürgen Herre , Bernhard Grill , Rainer Buchta , Karlheinz Brandenburg , Heinz Gerhäuser
发明人： Johannes Hilpert , Jürgen Herre , Bernhard Grill , Rainer Buchta , Karlheinz Brandenburg , Heinz Gerhäuser
IPC分类号： G10L1900
CPC分类号： H04B1/665
摘要： A method for detecting a transient in a discrete-time audio signal is performed completely in the time domain and includes the step of segmenting the discrete-time audio signal so as to generate consecutive segments of the same length with unfiltered discrete-time audio signals xs(T−1). The discrete-time audio signal in a current segment is subsequently filtered. Then either the energy of the filtered discrete-time audio signal in the current segment can be compared with the energy of the filtered discrete-time audio signal in a preceding segment or a current relationship between the energy of the filtered discrete-time audio signal in the current segment and the energy of the unfiltered discrete-time audio signal in the current segment can be formed and this current relationship compared with a preceding corresponding relationship. On the basis of the one and/or the other of these comparisons it is detected whether a transient is present in the discrete-time audio signal.
摘要翻译：用于检测离散时间音频信号中的瞬态的方法在时域中完全执行，并且包括分段离散时间音频信号以便生成具有未滤波的离散时间音频信号xs的相同长度的连续片段的步骤（T-1）。随后过滤当前片段中的离散时间音频信号。然后可以将当前段中滤波的离散时间音频信号的能量与前一段中滤波的离散时间音频信号的能量或滤波后的离散时间音频信号的能量之间的当前关系进行比较可以形成当前段的当前段和未过滤离散时间音频信号的能量，并将该当前关系与先前的对应关系进行比较。基于这些比较中的一个和/或另一个，检测离散时间音频信号中是否存在瞬态。

48. 发明授权

US08321207B2 Device and method for postprocessing spectral values and encoder and decoder for audio signals 有权
标题翻译：用于后处理光谱值的装置和方法以及用于音频信号的编码器和解码器
公开(公告)号：US08321207B2
公开(公告)日：2012-11-27
申请号：US12446772
申请日：2007-09-28
申请人： Bernd Edler , Ralf Geiger , Christian Ertel , Johannes Hilpert , Harald Popp
发明人： Bernd Edler , Ralf Geiger , Christian Ertel , Johannes Hilpert , Harald Popp
IPC分类号： G10L19/02 , G10L19/14
CPC分类号： G10L19/26 , G10L19/0017 , G10L19/24 , G10L19/265
摘要： For postprocessing spectral values which are based on a first transformation algorithm for converting the audio signal into a spectral representation, first a sequence of blocks of the spectral values representing a sequence of blocks of samples of the audio signal are provided. Hereupon, a weighted addition of spectral values of the sequence of blocks of spectral values is performed in order to obtain a sequence of blocks of postprocessed spectral values, wherein the combination is performed such that for calculating a postprocessed spectral value for a frequency band and a time duration a spectral value of the sequence of blocks for the frequency band and the time duration and a spectral value for another frequency band or another time duration are used, wherein the combination is further performed such that such weighting factors are used that the postprocessed spectral values are an approximation to the spectral values as they are obtained by converting the audio signal into a spectral representation using a second transformation algorithm which is different from the first transformation algorithm. The postprocessed spectral values are in particular used for a difference formation within a scalable encoder or for an addition within a scalable decoder, respectively.
摘要翻译：对于基于用于将音频信号转换成频谱表示的第一变换算法的后处理频谱值，首先提供表示音频信号的采样块序列的频谱值的块序列。因此，执行频谱值块序列的频谱值的加权相加以获得后处理频谱值的块序列，其中执行组合使得用于计算频带和后处理频谱值的后处理频谱值时间持续时间使用用于频带的块序列的频谱值和持续时间以及另一频带或另一持续时间的频谱值，其中进一步执行组合，使得使用这些加权因子后处理的频谱值是对频谱值的近似，因为它们是通过使用不同于第一变换算法的第二变换算法将音频信号转换成频谱表示而获得的。后处理的频谱值特别用于可分级编码器内的差分形成或分别用于可扩展解码器内的加法。

49. 发明申请

US20120020499A1 UPMIXER, METHOD AND COMPUTER PROGRAM FOR UPMIXING A DOWNMIX AUDIO SIGNAL 有权
标题翻译： UPMIXER，用于UPMIXING DOWNMIX音频信号的方法和计算机程序
公开(公告)号：US20120020499A1
公开(公告)日：2012-01-26
申请号：US13191236
申请日：2011-07-26
申请人： Matthias Neusinger , Julien Robilliard , Johannes Hilpert
发明人： Matthias Neusinger , Julien Robilliard , Johannes Hilpert
IPC分类号： H04B1/00
CPC分类号： G10L19/008
摘要： An upmixer for upmixing a downmix audio signal into an upmixed audio signal describing one or more upmixed audio channels includes a parameter applier configured to apply upmixing parameters to upmix the downmix audio signal in order to obtain the upmixed audio signal. The parameter applier is configured to apply a phase shift to the downmix audio signal to obtain a phase-shifted version of the downmix audio signal, while leaving a decorrelated signal unmodified by the phase shift. The parameter applier is further configured to combine the phase-shifted version of the downmix audio signal with the decorrelated signal to obtain the upmixed audio signal.
摘要翻译：用于将下混合音频信号上混合到描述一个或多个上混音频通道的上混合音频信号的上混混器包括配置用于施加上混参数以混合下混音频信号以便获得上混音频信号的参数施加器。参数施加器被配置为将相移应用于下混音频信号以获得下混音频信号的相移版本，同时保留未被相移修改的解相关信号。参数施加器还被配置为将降混音频信号的相移版本与解相关信号组合以获得上混音频信号。

50. 发明申请

US20110282675A1 Apparatus and Method for Generating a Synthesis Audio Signal and for Encoding an Audio Signal 有权
标题翻译：用于产生合成音频信号和用于编码音频信号的装置和方法
公开(公告)号：US20110282675A1
公开(公告)日：2011-11-17
申请号：US13107687
申请日：2011-05-13
申请人： Frederik Nagel , Markus Multrus , Jeremie Lecomte , Stefan Bayer , Guillaume Fuchs , Johannes Hilpert , Julien Robilliard
发明人： Frederik Nagel , Markus Multrus , Jeremie Lecomte , Stefan Bayer , Guillaume Fuchs , Johannes Hilpert , Julien Robilliard
IPC分类号： G10L19/00
CPC分类号： G10L19/24 , G10L21/038
摘要： An apparatus for generating a synthesis audio signal using a patching control signal has a first converter, a spectral domain patch generator, a high frequency reconstruction manipulator and a combiner. The first converter is configured for converting a time portion of an audio signal into a spectral representation. The spectral domain patch generator is configured for performing a plurality of different spectral domain patching algorithms, wherein each patching algorithm generates a modified spectral representation having spectral components in an upper frequency band derived from corresponding spectral components in a core frequency band of the audio signal. The spectral domain patch generator is furthermore configured to select a first spectral domain patching algorithm from the plurality of patching algorithms for a first time portion and a second spectral domain patching algorithm from the plurality of patching algorithm for a second different time portion in accordance with the patching control signal to obtain the modified spectral representation.
摘要翻译：使用补丁控制信号产生合成音频信号的装置具有第一转换器，频域补丁发生器，高频重建操纵器和组合器。第一转换器被配置为将音频信号的时间部分转换为频谱表示。频域补丁生成器被配置用于执行多个不同的频域修补算法，其中每个修补算法生成具有从音频信号的核心频带中的相应频谱分量导出的上频带中的频谱分量的修改频谱表示。频域补丁生成器还被配置为根据来自第一时间部分的多个修补算法和来自针对第二不同时间部分的多个修补算法的第二频域修补算法来选择第一频域修补算法修补控制信号以获得修改的光谱表示。

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