公开(公告)号：US07181079B2

公开(公告)日：2007-02-20

申请号：US10220651

申请日：2001-02-16

申请人： Juergen Herre ,  Martin Dietz ,  Thomas Sporer ,  Michael Schug ,  Wolfgang Schildbach

发明人： Juergen Herre ,  Martin Dietz ,  Thomas Sporer ,  Michael Schug ,  Wolfgang Schildbach

IPC分类号： G06K9/36 ,  H04B1/66 ,  H04N7/12 ,  H04N11/02 ,  H04N11/04 ,  G10L19/00 ,  G10L21/00

CPC分类号： H04N19/00 ,  H04N19/126 ,  H04N19/40 ,  H04N19/60

摘要： Analyzing an analysis time signal that has been generated from encoding and decoding and original time signal according to an encoding algorithm. The encoding block raster underlying the analysis time signal used by the encoding algorithm is determined. The analysis time signal is converted from its timely representation of analysis spectral coefficients to a spectral representation by using the established encoding block raster. At least two analysis spectral coefficients are grouped. The greatest common divisor of the analysis spectral coefficients are calculated, corresponding to the quantization step width used when quantizing the encoding algorithm or an integer multiple of it. In the case of an audio signal, the scale factor can easily be established for this group of spectral coefficients, i.e., for a scale factor band, from the quantization step width. All parameters used for the quantization of the original time signal are known; full iteration loops need not be performed.

摘要翻译： 根据编码算法分析从编码和解码产生的分析时间信号和原始时间信号。 确定由编码算法使用的分析时间信号的编码块光栅。 分析时间信号通过使用建立的编码块光栅从分析频谱系数的及时表示转换为频谱表示。 至少两个分析光谱系数被分组。 计算分析频谱系数的最大公约数，对应于量化编码算法时使用的量化步长或其整数倍。 在音频信号的情况下，从量化步长可以容易地为该组频谱系数（即，比例因子频带）建立比例因子。 用于量化原始时间信号的所有参数是已知的; 不需要执行完整的迭代循环。

公开(公告)号：US20060020958A1

公开(公告)日：2006-01-26

申请号：US10931635

申请日：2004-08-31

申请人： Eric Allamanche ,  Juergen Herre ,  Oliver Hellmuth ,  Thorsten Kastner ,  Markus Cremer

发明人： Eric Allamanche ,  Juergen Herre ,  Oliver Hellmuth ,  Thorsten Kastner ,  Markus Cremer

IPC分类号： H04H9/00

CPC分类号： G10L25/48

摘要： An apparatus for producing a fingerprint signal from an audio signal includes a means for calculating energy values for frequency bands of segments of the audio signal which are successive in time, so as to obtain, from the audio signal, a sequence of vectors of energy values, a means for scaling the energy values to obtain a sequence of scaled vectors, and a means for temporal filtering of the sequence of scaled vectors to obtain a filtered sequence which represents the fingerprint, or from which the fingerprint may be derived. Thus, a fingerprint is produced which is robust against disturbances due to problems associated with coding or with transmission channels, and which is especially suited for mobile radio applications.

公开(公告)号：US06750789B2

公开(公告)日：2004-06-15

申请号：US10168456

申请日：2002-10-25

申请人： Juergen Herre ,  Karlheinz Brandenburg ,  Thomas Sporer ,  Michael Schug ,  Wolfgang Schildbach

发明人： Juergen Herre ,  Karlheinz Brandenburg ,  Thomas Sporer ,  Michael Schug ,  Wolfgang Schildbach

IPC分类号： H03M700

CPC分类号： G10L19/02

摘要： In determining a coding block raster on which a decoded signal is based, a segment of the decoded signal is picked out first, said segment beginning at a certain output sampling value of the decoded signal. Said segment is then converted into a spectral representation, whereupon said spectral representation is then evaluated in relation to a predetermined criterion in order to obtain an evaluation result for the segment. This procedure is repeated for a plurality of different segments beginning at different output sampling values each, in order to obtain a plurality of evaluation results. Finally, the plurality of the evaluation results is searched in order to establish the evaluation result that has an extreme value as compared to the other evaluation results, in such a way that it can be assumed that the segment to which this evaluation result is allocated matches the coding block raster on which the decoded signal is based. This method can be used to determine the coding block raster for any decoded signal that has no explicit information about its coding block raster.

摘要翻译： 在确定解码信号所基于的编码块光栅时，首先选择解码信号的片段，所述片段以解码信号的特定输出采样值开始。 然后将所述段转换为频谱表示，然后根据预定标准对所述频谱表示进行评估，以获得该段的评估结果。 对于从不同的输出采样值开始的多个不同的段重复该过程，以获得多个评估结果。 最后，搜索多个评估结果以便建立与其他评估结果相比具有极值的评估结果，使得可以假定分配了该评估结果的分段匹配 解码信号所基于的编码块光栅。 该方法可以用于确定任何没有关于其编码块光栅的显式信息的解码信号的编码块栅格。

公开(公告)号：US09484038B2

公开(公告)日：2016-11-01

申请号：US13445585

申请日：2012-04-12

申请人： Giovanni Del Galdo ,  Oliver Thiergart ,  Juergen Herre ,  Fabian Kuech ,  Emanuel Habets ,  Alexandra Craciun ,  Achim Kuntz

发明人： Giovanni Del Galdo ,  Oliver Thiergart ,  Juergen Herre ,  Fabian Kuech ,  Emanuel Habets ,  Alexandra Craciun ,  Achim Kuntz

IPC分类号： H04R5/00 ,  G10L19/008 ,  G10L21/0216 ,  G06K9/62 ,  G10L21/0272

CPC分类号： G10L19/008 ,  G06K9/624 ,  G10L21/0272 ,  G10L2021/02166 ,  H04R5/00 ,  H04S2420/03 ,  H04S2420/11

摘要： An apparatus for generating a merged audio data stream is provided. The apparatus includes a demultiplexer for obtaining a plurality of single-layer audio data streams, wherein each input audio data stream includes one or more layers, wherein the demultiplexer is adapted to demultiplex each one of one or more input audio data streams having one or more layers into two or more demultiplexed audio data streams having exactly one layer. Furthermore, the apparatus includes a merging module for generating the merged audio data stream based on the plurality of single-layer audio data streams. Each layer of the input data audio streams, of the demultiplexed audio data streams, of the single-layer data streams and of the merged audio data stream includes a pressure value of a pressure signal, a position value and a diffuseness value as audio data.

摘要翻译： 提供了一种用于生成合并的音频数据流的装置。 该装置包括用于获得多个单层音频数据流的解复用器，其中每个输入音频数据流包括一个或多个层，其中，解复用器适于解复用一个或多个输入音频数据流中的每一个具有一个或多个 层分成两个或更多个具有恰好一层的解复用音频数据流。 此外，该装置包括用于基于多个单层音频数据流生成合并音频数据流的合并模块。 单层数据流和合并音频数据流的解复用的音频数据流的输入数据音频流的每一层包括作为音频数据的压力信号，位置值和扩散度值的压力值。

公开(公告)号：US09357305B2

公开(公告)日：2016-05-31

申请号：US13592977

申请日：2012-08-23

申请人： Fabian Kuech ,  Juergen Herre ,  Christof Faller ,  Christophe Tournery

发明人： Fabian Kuech ,  Juergen Herre ,  Christof Faller ,  Christophe Tournery

IPC分类号： H04R5/00 ,  G10L19/008 ,  G10L19/26 ,  G10L21/02

CPC分类号： H04R5/00 ,  G10L19/008 ,  G10L19/265 ,  G10L21/02

摘要： An apparatus for generating an enhanced downmix signal on the basis of a multi-channel microphone signal has a spatial analyzer configured to compute a set of spatial cue parameters having a direction information describing a direction-of-arrival of a direct sound, a direct sound power information and a diffuse sound power information on the basis of the multi-channel microphone signal. The apparatus also has a filter calculator for calculating enhancement filter parameters in dependence on the direction information describing the direction-of-arrival of the direct sound, in dependence on the direct sound power information and in dependence on the diffuse sound power information. The apparatus also has a filter for filtering the microphone signal, or a signal derived therefrom, using the enhancement filter parameters, to obtain the enhanced downmix signal.

摘要翻译： 一种用于基于多通道麦克风信号产生增强的下混合信号的装置具有空间分析器，该空间分析器被配置为计算具有描述直接声音到达方向的方向信息的一组空间提示参数，直接声音 功率信息和基于多声道麦克风信号的漫射声功率信息。 该装置还具有滤波器计算器，用于根据直接声功率信息和依赖于扩散声功率信息，根据描述直接声音到达方向的方向信息来计算增强滤波器参数。 该装置还具有用于使用增强滤波器参数对麦克风信号进行滤波的滤波器或从其导出的信号，以获得增强的降混信号。

公开(公告)号：US20120143613A1

公开(公告)日：2012-06-07

申请号：US13284583

申请日：2011-10-28

申请人： Juergen Herre ,  Andreas Hoelzer ,  Leonid Terentiev ,  Thorsten Kastner ,  Cornella Falch ,  Heiko Purnhagen ,  Jonas Engdegard ,  Falko Ridderbusch

发明人： Juergen Herre ,  Andreas Hoelzer ,  Leonid Terentiev ,  Thorsten Kastner ,  Cornella Falch ,  Heiko Purnhagen ,  Jonas Engdegard ,  Falko Ridderbusch

IPC分类号： G10L19/00

CPC分类号： G10L19/008 ,  G10L19/20

摘要： An apparatus for providing one or more adjusted parameters for a provision of an upmix signal representation on the basis of a downmix signal representation and an object-related parametric information includes a parameter adjuster. The parameter adjuster is configured to receive one or more input parameters and to provide, on the basis thereof, one or more adjusted parameters. The parameter adjuster is configured to provide the one or more adjusted parameters in dependence on the one or more input parameters and the object-related parametric information, such that a distortion of the upmix signal representation caused by the use of non-optimal parameters is reduced at least for input parameters deviating from optimal parameters by more than a predetermined deviation.

摘要翻译： 基于下混合信号表示和对象相关参数信息，提供用于提供上混合信号表示的一个或多个调整参数的装置包括参数调整器。 参数调整器被配置为接收一个或多个输入参数并且在其基础上提供一个或多个调整的参数。 参数调整器被配置为根据一个或多个输入参数和对象相关参数信息来提供一个或多个经调整的参数，使得由使用非最优参数引起的上混信号表示的失真减少 至少对于偏离最佳参数的输入参数大于预定偏差。

公开(公告)号：US20120114126A1

公开(公告)日：2012-05-10

申请号：US13289252

申请日：2011-11-04

申请人： Oliver Thiergart ,  Cornelia Falch ,  Fabian Kuech ,  Giovanni Del Galdo ,  Juergen Herre ,  Markus Kallinger

发明人： Oliver Thiergart ,  Cornelia Falch ,  Fabian Kuech ,  Giovanni Del Galdo ,  Juergen Herre ,  Markus Kallinger

IPC分类号： H04R5/00

CPC分类号： G10L21/0272 ,  G10L19/008

摘要： An audio format transcoder for transcoding an input audio signal, the input audio signal having at least two directional audio components. The audio format transcoder including a converter for converting the input audio signal into a converted signal, the converted signal having a converted signal representation and a converted signal direction of arrival. The audio format transcoder further includes a position provider for providing at least two spatial positions of at least two spatial audio sources and a processor for processing the converted signal representation based on the at least two spatial positions to obtain at least two separated audio source measures.

摘要翻译： 一种用于对输入音频信号进行代码转换的音频格式转码器，所述输入音频信号具有至少两个方向音频分量。 音频格式转码器包括用于将输入音频信号转换成转换信号的转换器，转换后的信号具有转换的信号表示和转换的信号到达方向。 音频格式转码器还包括用于提供至少两个空间音频源的至少两个空间位置的位置提供器和用于基于至少两个空间位置处理转换的信号表示的处理器，以获得至少两个分离的音频源测量。

公开(公告)号：US07903824B2

公开(公告)日：2011-03-08

申请号：US11032689

申请日：2005-01-10

申请人： Christof Faller ,  Juergen Herre

发明人： Christof Faller ,  Juergen Herre

IPC分类号： H04R5/00

CPC分类号： G10L19/008

摘要： At an audio encoder, cue codes are generated for one or more audio channels, wherein a combined cue code (e.g., a combined inter-channel correlation (ICC) code) is generated by combining two or more estimated cue codes, each estimated cue code estimated from a group of two or more channels. At an audio decoder, E transmitted audio channel(s) are decoded to generate C playback audio channels. Received cue codes include a combined cue code (e.g., a combined ICC code). One or more transmitted channel(s) are upmixed to generate one or more upmixed channels. One or more playback channels are synthesized by applying the cue codes to the one or more upmixed channels, wherein two or more derived cue codes are derived from the combined cue code, and each derived cue code is applied to generate two or more synthesized channels.

摘要翻译： 在音频编码器处，为一个或多个音频通道生成提示码，其中通过组合两个或多个估计的提示码来生成组合的提示码（例如，组合的信道间相关（ICC）码），每个估计的提示码 从一组两个或多个渠道估计。 在音频解码器处，E个发送的音频信道被解码以产生C个播放音频信道。 接收的提示码包括组合的提示码（例如，组合的ICC码）。 一个或多个传输的信道被混合以产生一个或多个上混频道。 通过将提示码应用于一个或多个上混合信道来合成一个或多个播放频道，其中从组合提示码导出两个或更多个派生提示码，并且应用每个派生的提示码以生成两个或多个合成频道。

公开(公告)号：US20100241433A1

公开(公告)日：2010-09-23

申请号：US12305936

申请日：2007-05-16

申请人： Juergen Herre ,  Bernhard Grill ,  Markus Multrus ,  Stefan Bayer ,  Ulrich Kraemer ,  Jens Hirschfeld ,  Stefan Wabnik ,  Gerald Schuller

发明人： Juergen Herre ,  Bernhard Grill ,  Markus Multrus ,  Stefan Bayer ,  Ulrich Kraemer ,  Jens Hirschfeld ,  Stefan Wabnik ,  Gerald Schuller

IPC分类号： G10L19/00

CPC分类号： G10L19/22 ,  G10L19/022

摘要： An audio encoder, an audio decoder or an audio processor includes a filter (12) for generating a filtered audio signal, the filter having a variable warping characteristic, the characteristic being controllable in response to a time-varying control signal (16), the control signal indicating a small or no warping characteristic or a comparatively high warping characteristic. Furthermore, a controller (18) is connected for providing the time-varying control signal, which depends on the audio signal. The filtered audio signal can be introduced to an encoding processor (22) having different encoding algorithms, one of which is a coding algorithm adapted to a specific signal pattern. Alternatively, the filter is a post-filter receiving a decoded audio signal.

摘要翻译： 音频编码器，音频解码器或音频处理器包括用于产生经滤波的音频信号的滤波器（12），该滤波器具有可变翘曲特性，该特性可响应于时变控制信号（16）而被控制， 指示变形特性小或不翘曲特性的控制信号。 此外，控制器（18）被连接用于提供取决于音频信号的时变控制信号。 滤波后的音频信号可以被引入具有不同编码算法的编码处理器（22），其中之一是适用于特定信号模式的编码算法。 或者，滤波器是接收经解码的音频信号的后置滤波器。

公开(公告)号：US07720230B2

公开(公告)日：2010-05-18

申请号：US11006482

申请日：2004-12-07

申请人： Eric Allamanche ,  Sascha Disch ,  Christof Faller ,  Juergen Herre

发明人： Eric Allamanche ,  Sascha Disch ,  Christof Faller ,  Juergen Herre

IPC分类号： H04R5/00

CPC分类号： G10L19/008

摘要： At an audio encoder, cue codes are generated for one or more audio channels, wherein an envelope cue code is generated by characterizing a temporal envelope in an audio channel. At an audio decoder, E transmitted audio channel(s) are decoded to generate C playback audio channels, where C>E≧1. Received cue codes include an envelope cue code corresponding to a characterized temporal envelope of an audio channel corresponding to the transmitted channel(s). One or more transmitted channel(s) are upmixed to generate one or more upmixed channels. One or more playback channels are synthesized by applying the cue codes to the one or more upmixed channels, wherein the envelope cue code is applied to an upmixed channel or a synthesized signal to adjust a temporal envelope of the synthesized signal based on the characterized temporal envelope such that the adjusted temporal envelope substantially matches the characterized temporal envelope.

摘要翻译： 在音频编码器中，为一个或多个音频通道生成提示码，其中通过表征音频通道中的时间包络来产生包络线索码。 在音频解码器处，对E个发送的音频信道进行解码以生成C个回放音频信道，其中C>E≥1。 接收的提示码包括与对应于所发送的频道的音频信道的特征化时间包络对应的信封提示码。 一个或多个传输的信道被混合以产生一个或多个上混频道。 通过将提示码应用于一个或多个上混合通道来合成一个或多个回放通道，其中，将包络提示码应用于上混合通道或合成信号，以基于表征的时间包络线来调整合成信号的时间包络 使得调整的时间包络基本上与所表征的时间包络相匹配。