公开(公告)号：US09462404B2

公开(公告)日：2016-10-04

申请号：US13588139

申请日：2012-08-17

申请人： Jürgen Herre ,  Johannes Hilpert ,  Stefan Geyersberger ,  Andreas Hölzer ,  Claus Spenger

发明人： Jürgen Herre ,  Johannes Hilpert ,  Stefan Geyersberger ,  Andreas Hölzer ,  Claus Spenger

IPC分类号： G10L19/00 ,  H04S3/00 ,  H04S3/02 ,  G10L19/008

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, 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.

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。 对于所选择的原始信道，计算信道侧信息，使得当使用信道侧信息加权时，包括第一和第二下混通道的下混通道或组合下混通道导致所选原始通道的近似。 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。

公开(公告)号：US09043200B2

公开(公告)日：2015-05-26

申请号：US12948417

申请日：2010-11-17

申请人： Ralph Sperschneider ,  Jürgen Herre ,  Karsten Linzmeier ,  Johannes Hilpert

发明人： Ralph Sperschneider ,  Jürgen Herre ,  Karsten Linzmeier ,  Johannes Hilpert

IPC分类号： G10L19/00 ,  G10L21/00 ,  G06K9/36 ,  H03K7/00 ,  H03M7/40 ,  G10L19/008 ,  H04N19/196 ,  H04N19/15 ,  H04N19/12 ,  H04N19/463 ,  H04N19/91 ,  G10L19/02 ,  G10L19/002 ,  G10L25/18

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 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 a second sequence of tuples. The first sequence of tuples includes tuples of parameters having two parameters from a single portion of the original signal and the second sequence of tuples includes tuples of parameters having one parameter from the first portion and one parameter from the second portion of the original signal. A bit estimator estimates the number of necessary bits to encode the first and the second sequence of tuples. Only the sequence of tuples, which results in the lower number of bits, is encoded.

摘要翻译： 本发明基于以下发现：包括：原始信号的第一部分的表示的第一组参数和原始信号的第二部分的表示的第二组参数的参数可以被有效地编码， 参数以元组的第一序列和第二序列的元组排列。 元组的第一序列包括具有来自原始信号的单个部分的两个参数的参数的元组，并且第二序列元组包括具有来自第一部分的一个参数的参数元组和来自原始信号的第二部分的一个参数。 比特估计器估计要对第一和第二个元组序列进行编码的必需比特数。 只有导致较低位数的元组序列被编码。

公开(公告)号：US20090304198A1

公开(公告)日：2009-12-10

申请号：US12296698

申请日：2007-03-28

申请人： Jürgen Herre ,  Herbert Buchner

发明人： Jürgen Herre ,  Herbert Buchner

IPC分类号： H04B3/20

CPC分类号： H04M9/082 ,  G10L19/008 ,  G10L2021/02082

摘要： An audio signal decorrelator for deriving an output audio signal from an input audio signal has a frequency analyzer for extracting from the input audio signal a first partial signal descriptive of an audio content in a first audio frequency range and a second partial signal descriptive of an audio content in a second audio frequency range having higher frequencies compared to the second audio frequency range. A partial signal modifier for modifies the first and second partial signals, to obtain first and second processed partial signals, so that a modulation amplitude of a time variant phase shift or time variant delay applied to the first partial signal is higher than that applied to the second partial signal, or for modifying only the first partial signal. A signal combiner combines the first and second processed partial signals, or combines the first processed partial signal and the second partial signal, to obtain an output audio signal.

摘要翻译： 用于从输入音频信号导出输出音频信号的音频信号解相关器具有频率分析器，用于从输入音频信号中提取描述第一音频频率范围内的音频内容的第一部分信号和描述音频的第二部分信号 与第二音频范围相比具有较高频率的第二音频频率范围内的内容。 一种部分信号修改器，用于修改第一和第二部分信号，以获得第一和第二处理的部分信号，使得施加到第一部分信号的时变相移或时变延迟的调制幅度高于施加到第一部分信号的调制幅度 第二部分信号，或仅用于修改第一部分信号。 信号组合器组合第一和第二处理的部分信号，或组合第一处理部分信号和第二部分信号，以获得输出音频信号。

公开(公告)号：US06766293B1

公开(公告)日：2004-07-20

申请号：US09367775

申请日：1999-08-18

申请人： Jürgen Herre ,  Uwe Gbur ,  Andreas Ehret ,  Martin Dietz ,  Bodo Teichmann ,  Oliver Kunz ,  Karlheinz Brandenburg ,  Heinz Gerhäuser

发明人： Jürgen Herre ,  Uwe Gbur ,  Andreas Ehret ,  Martin Dietz ,  Bodo Teichmann ,  Oliver Kunz ,  Karlheinz Brandenburg ,  Heinz Gerhäuser

IPC分类号： G10L2102

CPC分类号： G10L19/028 ,  H04B1/665

摘要： In a method for signalling a noise substitution when coding an audio signal, the time-domain audio signal is first transformed into the frequency domain to obtain spectral values. The spectral values are subsequently grouped together to form groups of spectral values. On the basis of a detection establishing whether a group of spectral values is a noisy group or not, a codebook is allocated to a non-noisy or tonal group by means of a codebook number for redundancy coding of the same. If a group is noisy, an additional codebook number which does not refer to a codebook is allocated to it in order to signal that this group is noisy and therefore does not have to be redundancy coded. By signalling noise substitution by means of a Huffman codebook number for noisy groups of spectral values, which are e.g. sections made up of scale factor bands which do not have to be redundancy coded, an opportunity is provided to indicate the presence of a noise substitution in a scale factor band in the bit stream syntax of the MPEG-2 Advanced Audio Coding (AAC) Standard without having to interfere with the basic coding structure and without having to meddle with the structure of the existing bit stream syntax.

摘要翻译： 在对音频信号编码时用于发信号通知的方法中，首先将时域音频信号变换成频域以获得频谱值​​。 光谱值随后被分组在一起以形成光谱值组。 基于确定一组频谱值是否为噪声组的检测，通过用于冗余编码的码本号将码本分配给非噪声或色调组。 如果组噪声，则分配不附加码本的附加码本号，以便发信号通知该组噪声，因此不必进行冗余编码。 通过用于噪声组的频谱值的霍夫曼码本号对信号进行信号替换， 由不必冗余编码的比例因子带组成的部分提供了一种机会，以指示在MPEG-2高级音频编码（AAC）标准的比特流语法中的比例因子频带中存在噪声替换 而不必干扰基本编码结构，而不必介入现有比特流语法的结构。

公开(公告)号：US20130016843A1

公开(公告)日：2013-01-17

申请号：US13588139

申请日：2012-08-17

申请人： JÜRGEN HERRE ,  JOHANNES HILPERT ,  STEFAN GEYERSBERGER ,  ANDREAS HÖLZER ,  CLAUS SPENGER

发明人： JÜRGEN HERRE ,  JOHANNES HILPERT ,  STEFAN GEYERSBERGER ,  ANDREAS HÖLZER ,  CLAUS SPENGER

IPC分类号： 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, 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.

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。 对于所选择的原始信道，计算信道侧信息，使得当使用信道侧信息加权时，包括第一和第二下混通道的下混通道或组合下混通道导致所选原始通道的近似。 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。

公开(公告)号：US07983922B2

公开(公告)日：2011-07-19

申请号：US11212395

申请日：2005-08-25

申请人： Matthias Neusinger ,  Jürgen Herre ,  Sascha Disch ,  Heiko Purnhagen ,  Kristofer Kjörling ,  Jonas Engdegard ,  Jeroen Breebaart ,  Erik Schuijers ,  Werner Oomen

发明人： Matthias Neusinger ,  Jürgen Herre ,  Sascha Disch ,  Heiko Purnhagen ,  Kristofer Kjörling ,  Jonas Engdegard ,  Jeroen Breebaart ,  Erik Schuijers ,  Werner Oomen

IPC分类号： G10L19/00 ,  H04R5/00

CPC分类号： G10L19/26 ,  G10L19/008 ,  G10L2019/0012 ,  H04S3/008

摘要： On an encoder-side, a multi-channel input signal is analyzed for obtaining smoothing control information, which is to be used by a decoder-side multi-channel synthesis for smoothing quantized transmitted parameters or values derived from the quantized transmitted parameters for providing an improved subjective audio quality in particular for slowly moving point sources and rapidly moving point sources having tonal material such as fast moving sinusoids.

摘要翻译： 在编码器侧，分析多通道输入信号以获得平滑控制信息，该平滑控制信息将被解码器侧多信道合成用于平滑经量化的传输参数或从量化的传输参数导出的值，以提供 改善主观音频质量，特别是对于缓慢移动的点源和具有诸如快速移动的正弦波的音调材料的快速移动的点源。

公开(公告)号：US07447317B2

公开(公告)日：2008-11-04

申请号：US10679085

申请日：2003-10-02

申请人： Jürgen Herre ,  Johannes Hilpert ,  Stefan Geyersberger ,  Andreas Hölzer ,  Claus Spenger

发明人： Jürgen Herre ,  Johannes Hilpert ,  Stefan Geyersberger ,  Andreas Hölzer ,  Claus Spenger

IPC分类号： G10L19/00 ,  G10L11/00 ,  G10L21/02 ,  H04R5/00 ,  G06F17/00

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.

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。

公开(公告)号：US07012183B2

公开(公告)日：2006-03-14

申请号：US10713691

申请日：2003-11-14

申请人： Jürgen Herre ,  Jan Rohden ,  Christian Uhle ,  Markus Cremer

发明人： Jürgen Herre ,  Jan Rohden ,  Christian Uhle ,  Markus Cremer

IPC分类号： G10H1/40

CPC分类号： G10L19/0208 ,  G10H1/40 ,  G10H2210/076 ,  G10H2250/135 ,  G10L25/06 ,  G10L25/90

摘要： An apparatus for analyzing an audio signal with regard to rhythm information of the audio signal by using an autocorrelation function comprises a filter bank for separating the audio signal into at least two sub-band signals. The sub-band signals are examined with regard to periodicities by an autocorrelation function, to obtain rhythm raw-information for the at least two sub-band signals. To reduce or eliminate the ambiguities of the autocorrelation function for periodical signals, the rhythm raw-information is postprocessed to obtain post-processed rhythm raw-information for the sub-band signal. The rhythm information of the audio signal is established based on the postprocessed rhythm raw-information. By the sub-band-wise ACF postprocessing, ACF ambiguities are already eliminated where they originate, and rhythm portions are added at double tempi, which an autocorrelation function processing does normally not provide, so that, as a result, a more robust determination of the rhythm information of the audio signal arises.

摘要翻译： 用于通过使用自相关函数来分析关于音频信号的节奏信息的音频信号的装置包括用于将音频信号分离成至少两个子带信号的滤波器组。 通过自相关函数检查子带信号的周期性，以获得用于至少两个子带信号的节奏原始信息。 为了减少或消除周期信号的自相关函数的不确定性，后处理节奏原始信息以获得用于子带信号的后处理节奏原始信息。 基于后处理的节奏原始信息建立音频信号的节奏信息。 通过子带式ACF后处理，ACF模糊度已经被消除，它们起源，并且节奏部分以双重温度被添加，自相关函数处理通常不提供，因此，结果是更稳健地确定 音频信号的节奏信息出现。

公开(公告)号：US06424939B1

公开(公告)日：2002-07-23

申请号：US09402684

申请日：1999-10-06

申请人： Jürgen Herre ,  Uwe Gbur ,  Andreas Ehret ,  Martin Dietz ,  Bodo Teichmann ,  Oliver Kunz ,  Karlheinz Brandenburg ,  Heinz Gerhauser

发明人： Jürgen Herre ,  Uwe Gbur ,  Andreas Ehret ,  Martin Dietz ,  Bodo Teichmann ,  Oliver Kunz ,  Karlheinz Brandenburg ,  Heinz Gerhauser

IPC分类号： G10L1900

CPC分类号： H04B1/665 ,  G10L19/028

摘要： A method for coding or decoding an audio signal combines the advantages of TNS processing and noise substitution. A time-discrete audio signal is initially transformed to the frequency domain in order to obtain spectral values of the temporal audio signal. Subsequently, a prediction of the spectral values in relation to frequency is carried out in order to obtain spectral residual values. Within the spectral residual values, areas are detected encompassing spectral residual values with noise properties. The spectral residual values in the noise areas are noise-substituted, whereupon information concerning the noise areas and noise substitution is incorporated into side information pertaining to a coded audio signal. Thus, considerable bit savings in case of transient signals can be achieved.

摘要翻译： 用于对音频信号进行编码或解码的方法结合了TNS处理和噪声替换的优点。 时间离散音频信号最初被变换到频域以获得时间音频信号的频谱值。 随后，进行与频率相关的频谱值的预测，以获得谱残差值。 在光谱残差值内，检测到包含具有噪声特性的光谱残差值的区域。 噪声区域中的频谱残差值被噪声替代，因此关于噪声区域和噪声替换的信息被并入与编码音频信号有关的侧面信息中。 因此，可以实现在瞬态信号的情况下相当可观的位节省。

公开(公告)号：US08538037B2

公开(公告)日：2013-09-17

申请号：US12296698

申请日：2007-03-28

申请人： Jürgen Herre ,  Herbert Buchner

发明人： Jürgen Herre ,  Herbert Buchner

IPC分类号： H04B15/00 ,  H04M9/08 ,  H04M1/00 ,  H04M1/38

CPC分类号： H04M9/082 ,  G10L19/008 ,  G10L2021/02082

摘要： An audio signal decorrelator for deriving an output audio signal from an input audio signal has a frequency analyzer for extracting from the input audio signal a first partial signal descriptive of an audio content in a first audio frequency range and a second partial signal descriptive of an audio content in a second audio frequency range having higher frequencies compared to the second audio frequency range. A partial signal modifier modifies the first and second partial signals, to obtain first and second processed partial signals, so that a modulation amplitude of a time variant phase shift or time variant delay applied to the first partial signal is higher than that applied to the second partial signal, or for modifying only the first partial signal. A signal combiner combines the first and second processed partial signals, or combines the first processed partial signal and the second partial signal, to obtain an output audio signal.

摘要翻译： 用于从输入音频信号导出输出音频信号的音频信号解相关器具有频率分析器，用于从输入音频信号中提取描述第一音频频率范围内的音频内容的第一部分信号和描述音频的第二部分信号 与第二音频范围相比具有较高频率的第二音频频率范围内的内容。 部分信号修改器修改第一和第二部分信号，以获得第一和第二处理的部分信号，使得施加到第一部分信号的时变相移或时变延迟的调制幅度高于施加到第二部分信号的调制幅度 部分信号，或仅修改第一部分信号。 信号组合器组合第一和第二处理的部分信号，或组合第一处理部分信号和第二部分信号，以获得输出音频信号。