-
1.发明授权IMPROVED MAGNITUDE RESPONSE AND TEMPORAL ALIGNMENT IN PHASE VOCODER BASED BANDWIDTH EXTENSION FOR AUDIO SIGNALS 有权基于相位声码器的音频信号带宽扩展改进的幅度响应和时间对准
公开(公告)号：EP2545551B1
公开(公告)日：2017-10-04
申请号：EP11707156.3
申请日：2011-03-04
IPC分类号： G10L21/038 , G10L19/022 , G10L19/02 , G10L19/16 , G10L19/26
CPC分类号： G10L19/0208 , G10L19/022 , G10L19/16 , G10L19/26 , G10L21/038
摘要： An apparatus for generating a bandwidth extended audio signal from an input signal, includes a patch generator for generating one or more patch signals from the input signal, wherein the patch generator is configured for performing a time stretching of subband signals from an analysis filterbank, and wherein the patch generator further includes a phase adjuster for adjusting phases of the subband signals using a filterbank-channel dependent phase correction.
摘要翻译： 一种用于从输入信号生成带宽扩展音频信号的设备，包括：用于从输入信号生成一个或多个修补信号的修补生成器，其中修补生成器被配置为执行来自分析滤波器组的子带信号的时间展宽，以及 其中所述补丁生成器还包括相位调整器，用于使用滤波器组通道相关的相位校正来调整所述子带信号的相位。
-
2.发明授权AUDIO SIGNAL DECODER, AUDIO SIGNAL ENCODER, METHODS AND COMPUTER PROGRAM USING A SAMPLING RATE DEPENDENT TIME-WARP CONTOUR ENCODING 有权声音信号解码器，TONSIGNALCODIERER，方法和计算机程序与相关采样ZEITSPRUNG轮廓编码
公开(公告)号：EP2532001B1
公开(公告)日：2014-04-02
申请号：EP11707665.3
申请日：2011-03-09
发明人： BAYER, Stefan , BÄCKSTRÖM, Tom , GEIGER, Ralf , EDLER, Bernd , DISCH, Sascha , VILLEMOES, Lars
IPC分类号： G10L19/02 , G10L25/90 , G10L19/022
CPC分类号： G10L19/022 , G10L19/0212 , G10L25/90
摘要： An audio signal decoder configured to provide a decoded audio signal representation on the basis of an encoded audio signal representation including a sampling frequency information, an encoded time warp information and an encoded spectrum representation includes a time warp calculator and a warp decoder. The time warp calculator is configured to adapt a mapping rule for mapping codewords of the encoded time warp information onto decoded time warp values describing the decoded time warp information in dependence on the sampling frequency information. The warp decoder is configured to provide the decoded audio signal representation on the basis of the encoded spectrum representation and in dependence on the decoded time warp information.
-
3.发明公开AUDIO SIGNAL ENCODER, AUDIO SIGNAL DECODER, METHOD FOR ENCODING OR DECODING AN AUDIO SIGNAL USING AN ALIASING-CANCELLATION 审中-公开TONSIGNALCODIERER，声音信号的解码器，方法编码，解码的声音信号的手段混叠抑制
公开(公告)号：EP2491556A1
公开(公告)日：2012-08-29
申请号：EP10771705.0
申请日：2010-10-19
申请人： Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. , VoiceAge Corporation , Koninklijke Philips Electronics N.V. , Dolby International AB
发明人： BESSETTE, Bruno , NEUENDORF, Max , GEIGER, Ralf , GOURNAY, Philippe , LEFEBVRE, Roch , GRILL, Bernhard , LECOMTE, Jeremie , BAYER, Stefan , RETTELBACH, Nikolaus , VILLEMOES, Lars , SALAMI, Redwan , DEN BRINKER, Albertus C.
CPC分类号： G10L19/03 , G10L19/0212 , G10L19/20 , G10L2019/0008
摘要： An audio signal decoder includes a transform domain path configured to obtain a time-domain representation of a portion of an audio content on the basis of a first set of spectral coefficients, a representation of an aliasing-cancellation stimulus signal and a plurality of linear-prediction-domain parameters. The transform domain path applies a spectrum shaping to the first set of spectral coefficients to obtain a spectrally-shaped version thereof. The transform domain path obtains a time-domain representation of the audio content on the basis of the spectrally-shaped version of the first set of spectral coefficients. The transform domain path includes an aliasing-cancellation stimulus filter to filter the aliasing-cancellation stimulus signal in dependence on at least a subset of the linear-prediction-domain parameters. The transform domain path also includes a combiner configured to combine the time-domain representation of the audio content with an aliasing-cancellation synthesis signal to obtain an aliasing reduced time-domain signal.
摘要翻译： 音频信号解码器包括被配置为获得第一组频谱系数的基础上的音频内容的一部分的一个时域表示变换域路径，一个混叠消除激励信号的表示和直链的多元 预测域参数。 变换域路径适用的频谱成形，以所述第一组频谱系数以获得其频谱形版本。 变换域路径获得的第一组频谱系数的频谱形版本的基础上音频内容的时域表示。 变换域路径包括在混叠消除刺激滤波器中的依赖于过滤的至少所述线性预测域参数的子集的混叠消除刺激信号。 因此，该变换域路径包括被配置为与上混叠消除合成信号，以获得混叠减小的时域信号的音频内容的时域表示的组合器。
-
公开(公告)号：EP2082397B1
公开(公告)日：2011-12-28
申请号：EP07818758.0
申请日：2007-10-05
申请人： Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. , Dolby International AB , Koninklijke Philips Electronics N.V.
发明人： HILPERT, Johannes , LINZMEIER, Karsten , HERRE, Jürgen , SPERSCHNEIDER, Ralph , HÖLZER, Andreas , VILLEMOES, Lars , ENGDEGARD, Jonas , PURNHAGEN, Heiko , KJÖRLING, Kristofer , BREEBAART, Jeroen , OOMEN, Werner
CPC分类号： G10L19/173 , G10L19/008 , H04S2420/03
摘要： A parameter transformer generates level parameters, indicating an energy relation between a first and a second audio channel of a multi-channel audio signal associated to a multi-channel loudspeake configuration. The level parameter are generated based on object parameters for a plurality of audio objects associated to a down-mix channel, which is generated using object audio signals associated to the audio objects. The object parameters comprise an energy parameter indicating an energy of the object audio signal. To derive the coherence and the level parameters, a parameter generator is used, which combines the energy parameter and object rendering parameters, which depend on a desired rendering configuration.
-
5.发明授权AUDIO SIGNAL DECODER, AUDIO SIGNAL ENCODER, METHOD FOR DECODING AN AUDIO SIGNAL, METHOD FOR ENCODING AN AUDIO SIGNAL AND COMPUTER PROGRAM USING A PITCH-DEPENDENT ADAPTATION OF A CODING CONTEXT 有权声解码器，TONSIGNALCODIERER方法用于解码的声音信号，METHOD FOR CODING WITH A CODIERKONTEXTES的音高相关适应的音频信号和计算机程序
公开(公告)号：EP2539893B1
公开(公告)日：2014-04-02
申请号：EP11707415.3
申请日：2011-03-09
发明人： BAYER, Stefan , BÄCKSTRÖM, Tom , GEIGER, Ralf , EDLER, Bernd , DISCH, Sascha , VILLEMOES, Lars
IPC分类号： G10L19/02 , G10L19/022 , G10L25/90
CPC分类号： G10L19/022 , G10L19/0212 , G10L25/90
摘要： An audio signal decoder configured to provide a decoded audio signal representation on the basis of an encoded audio signal representation including a sampling frequency information, an encoded time warp information and an encoded spectrum representation includes a time warp calculator and a warp decoder. The time warp calculator is configured to adapt a mapping rule for mapping codewords of the encoded time warp information onto decoded time warp values describing the decoded time warp information in dependence on the sampling frequency information. The warp decoder is configured to provide the decoded audio signal representation on the basis of the encoded spectrum representation and in dependence on the decoded time warp information.
-
6.发明公开AUDIO SIGNAL DECODER, AUDIO SIGNAL ENCODER, METHODS AND COMPUTER PROGRAM USING A SAMPLING RATE DEPENDENT TIME-WARP CONTOUR ENCODING 有权声音信号解码器，TONSIGNALCODIERER，方法和计算机程序与相关采样ZEITSPRUNG轮廓编码
公开(公告)号：EP2532001A1
公开(公告)日：2012-12-12
申请号：EP11707665.3
申请日：2011-03-09
发明人： BAYER, Stefan , BÄCKSTRÖM, Tom , GEIGER, Ralf , EDLER, Bernd , DISCH, Sascha , VILLEMOES, Lars
IPC分类号： G10L19/02
CPC分类号： G10L19/022 , G10L19/0212 , G10L25/90
摘要： An audio signal decoder configured to provide a decoded audio signal representation on the basis of an encoded audio signal representation including a sampling frequency information, an encoded time warp information and an encoded spectrum representation includes a time warp calculator and a warp decoder. The time warp calculator is configured to adapt a mapping rule for mapping codewords of the encoded time warp information onto decoded time warp values describing the decoded time warp information in dependence on the sampling frequency information. The warp decoder is configured to provide the decoded audio signal representation on the basis of the encoded spectrum representation and in dependence on the decoded time warp information.
-
公开(公告)号：EP3570278A1
公开(公告)日：2019-11-20
申请号：EP19179788.5
申请日：2011-03-04
IPC分类号： G10L19/02 , G10L21/04 , G10L21/038
摘要： An apparatus for processing an input audio signal (2300) relies on a cascade of filterbanks, the cascade comprising a synthesis filterbank (2304) for synthesizing an audio intermediate signal (2306) from the input audio signal (2300), the input audio signal being represented by a plurality of first subband signals (2303) generated by an analysis filterbank (2302), wherein a number of filterbank channels of the synthesis filterbank (2304) is smaller than a number of channels of the analysis filterbank (2302). The apparatus furthermore comprises a further analysis filterbank (2307) for generating a plurality of second subband signals (2308) from the audio intermediate signal (2306), wherein the further analysis filterbank has a number of channels being different from the number of channels of the synthesis filterbank (2304), so that a sampling rate of a subband signal of the plurality of second subband signals (2308) is different from a sampling rate of a first subband signal of the plurality of first subband signals (2303).
-
公开(公告)号：EP2543038B1
公开(公告)日：2015-08-26
申请号：EP11709735.2
申请日：2011-03-23
发明人： PURNHAGEN, Heiko , CARLSSON, Pontus , VILLEMOES, Lars , ROBILLARD, Julien , NEUSINGER, Matthias , HELMRICH, Christian , HILPERT, Johannes , RETTELBACH, Nikolaus , DISCH, Sascha , EDLER, Bernd
IPC分类号： G10L19/008 , G10L19/04
CPC分类号： G10L19/04 , G10L19/008
摘要： An audio encoder and an audio decoder are based on a combination of two audio channels (201, 202) to obtain a first combination signal (204) as a mid signal and a residual signal (205) which can be derived using a predicted side signal derived from the mid signal. The first combination signal and the prediction residual signal are encoded (209) and written (212) into a data stream (213) together with the prediction information (206) derived by an optimizer (207) based on an optimization target (208). A decoder uses the prediction residual signal, the first combination signal and the prediction information to derive a decoded first channel signal and a decoded second channel signal. In an encoder example or in a decoder example, a real-to-imaginary transform can be applied for estimating the imaginary part of the spectrum of the first combination signal. For calculating the prediction signal used in the derivation of the prediction residual signal, the real-valued first combination signal is multiplied by a real portion of the complex prediction information and the estimated imaginary part of the first combination signal is multiplied by an imaginary portion of the complex prediction information.
-
9.发明授权APPARATUS AND METHOD FOR PROCESSING AN AUDIO SIGNAL USING PATCH BORDER ALIGNMENT 有权设备和方法处理的音频信号片边界取向
公开(公告)号：EP2545553B1
公开(公告)日：2014-07-30
申请号：EP11715452.6
申请日：2011-03-04
申请人： Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. , Dolby International AB , Disch, Sascha
IPC分类号： G10L21/038 , G10L21/04 , G10L19/02
CPC分类号： G10L19/008 , G10L19/0204 , G10L21/038 , G10L21/04
摘要： An apparatus for processing an input audio signal relies on a cascade of filterbanks, the cascade having a synthesis filterbank for synthesizing an audio intermediate signal from the input audio signal, the input audio signal being represented by a plurality of first subband signals generated by an analysis filterbank, wherein a number of filterbank channels of the synthesis filterbank is smaller than a number of channels of the analysis filterbank. The apparatus furthermore has a further analysis filterbank for generating a plurality of second subband signals from the audio intermediate signal, wherein the further analysis filterbank has a number of channels being different from the number of channels of the synthesis filterbank, so that a sampling rate of a subband signal of the plurality of second subband signals is different from a sampling rate of a first subband signal of the plurality of first subband signals.
-
10.发明公开APPARATUS AND METHOD FOR PROCESSING AN AUDIO SIGNAL USING PATCH BORDER ALIGNMENT 有权设备和方法处理的音频信号片边界取向
公开(公告)号：EP2545553A1
公开(公告)日：2013-01-16
申请号：EP11715452.6
申请日：2011-03-04
申请人： Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. , Dolby International AB , Disch, Sascha
CPC分类号： G10L19/008 , G10L19/0204 , G10L21/038 , G10L21/04
摘要： An apparatus for processing an input audio signal relies on a cascade of filterbanks, the cascade having a synthesis filterbank for synthesizing an audio intermediate signal from the input audio signal, the input audio signal being represented by a plurality of first subband signals generated by an analysis filterbank, wherein a number of filterbank channels of the synthesis filterbank is smaller than a number of channels of the analysis filterbank. The apparatus furthermore has a further analysis filterbank for generating a plurality of second subband signals from the audio intermediate signal, wherein the further analysis filterbank has a number of channels being different from the number of channels of the synthesis filterbank, so that a sampling rate of a subband signal of the plurality of second subband signals is different from a sampling rate of a first subband signal of the plurality of first subband signals.
-
-
-
-
-
-
-
-
-
搜索结果
230 条记录 (耗时 0.029 秒)
