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71. 发明公开

EP2301024A1 AUDIO ENCODING/DECODING SCHEME HAVING A SWITCHABLE BYPASS 有权
标题翻译：与切换旁路音频编码/音频解码处理
公开(公告)号：EP2301024A1
公开(公告)日：2011-03-30
申请号：EP09797423.2
申请日：2009-07-06
申请人： Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. , VoiceAge Corporation
发明人： GRILL, Bernhard , BAYER, Stefan , FUCHS, Guillaume , GEYERSBERGER, Stefan , GEIGER, Ralf , HILPERT, Johannes , KRAEMER, Ulrich , LECOMTE, Jeremie , MULTRUS, Markus , NEUENDORF, Max , POPP, Harald , RETTELBACH, Nikolaus , LEFEBVRE, Roch , BESSETTE, Bruno , LAPIERRE, Jimmy , SALAMI, Redwan , GOURNAY, Philippe
IPC分类号： G10L19/14
CPC分类号： G10L19/173 , G10L19/0017 , G10L19/008 , G10L19/0212 , G10L19/18 , G10L2019/0008
摘要： An apparatus for encoding comprises a first domain converter (510), a switchable bypass (50), a second domain converter (410), a first processor (420) and a second processor (520) to obtain an encoded audio signal having different signal portions represented by coded data in different domains, which have been coded by different coding algorithms. Corresponding decoding stages in the decoder together with a bypass for bypassing a domain converter allow the generation of a decoded audio signal with high quality and low bit rate.

72. 发明公开

EP2260485A1 AUDIO SIGNAL DECODER, AUDIO SIGNAL ENCODER, ENCODED MULTI-CHANNEL AUDIO SIGNAL REPRESENTATION, METHODS AND COMPUTER PROGRAM 有权
标题翻译： AUDIOSIGNALDEKODIERER，AUDIOSIGNALKODIERER的，编码的多声道音频信号表示和方法和计算机程序THEREFOR
公开(公告)号：EP2260485A1
公开(公告)日：2010-12-15
申请号：EP09776910.3
申请日：2009-07-01
申请人： Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
发明人： BAYER, Stefan , DISCH, Sascha , GEIGER, Ralf , FUCHS, Guillaume , NEUENDORF, Max , SCHULLER, Gerald , EDLER, Bernd
IPC分类号： G10L19/00 , G10L19/02 , G10L21/04
CPC分类号： G10L21/04 , G10L19/0212 , G10L19/022 , G10L19/032 , G10L19/167
摘要： An audio signal decoder for providing a decoded multichannel audio signal representation on the basis of an encoded multi-channel audio signal representation comprises a time warp decoder configured to selectively use individual audio channel specific time warp contours or a joint multi-channel time warp contour for a reconstruction of a plurality of audio channels represented by the encoded multi-channel audio signal representation. An audio signal encoder for providing an encoded representation of a multichannel audio signal comprises an encoded audio representation provider configured to selectively provide an audio representation comprising a common time warp contour information, commonly associated with a plurality of audio channels of the multi-channel audio signal, or an encoded audio representation comprising individual time warp contour information, individually associated with the different audio channels of the plurality of audio channels, in dependence on an information describing a similarity or difference between time warp contours associated with the audio channels of the plurality of audio channels.

73. 发明授权

EP1964111B1 DEVICE AND METHOD FOR POSTPROCESSING SPECTRAL VALUES AND ENCODER AND DECODER FOR AUDIO SIGNALS 有权
标题翻译： DEVICE AND METHOD FOR后处理和频谱编码器解码器单元和用于音频信号
公开(公告)号：EP1964111B1
公开(公告)日：2010-11-24
申请号：EP07818557.6
申请日：2007-09-28
申请人： Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
发明人： EDLER, Bernd , GEIGER, Ralf , ERTEL, Christian , HILPERT, Johannes , POPP, Harald
IPC分类号： 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.

74. 发明公开

EP2147430A1 AUDIO TRANSFORM CODING USING PITCH CORRECTION 有权
标题翻译： AUDIO变换花边纠错编码
公开(公告)号：EP2147430A1
公开(公告)日：2010-01-27
申请号：EP09728768.4
申请日：2009-03-23
申请人： Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
发明人： EDLER, Bernd , DISCH, Sascha , GEIGER, Ralf , BAYER, Stefan , KRAEMER, Ulrich , FUCHS, Guillaume , NEUENDORF, Max , MULTRUS, Markus , SCHULLER, Gerald , POPP, Harald
IPC分类号： G10L19/02
CPC分类号： G10L19/0212 , G10L19/022
摘要： A processed representation of an audio signal having a sequence of frames is generated by sampling the audio signal within a first and a second frame of the sequence of frames, the second frame following the first frame, the sampling using information on a pitch contour of the first and the second frame to derive a first sampled representation. The audio signal is sampled within the second and the third frame, the third frame following the second frame in the sequence of frames. The sampling uses the information on the pitch contour of the second frame and information on a pitch contour of the third frame to derive a second sampled representation. A first scaling window is derived for the first sampled representation and a second scaling window is derived for the second sampled representation, the scaling windows depending on the samplings applied to derive the first sampled representations or the second sampled representation.

75. 发明授权

EP1604527B1 VORRICHTUNG UND VERFAHREN ZUM EINBETTEN EINER BINÄREN NUTZINFORMATION IN EIN TRÄGERSIGNAL 有权
标题翻译： DEVICE AND METHOD FOR AN嵌入BINARY有用的信息转换成载波信号
公开(公告)号：EP1604527B1
公开(公告)日：2006-11-15
申请号：EP04729869.0
申请日：2004-04-28
申请人： Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
发明人： SCHULLER, Gerald , GEIGER, Ralf , KOLLER, Jürgen
IPC分类号： H04N7/24 , H04N1/32 , G10L19/00
CPC分类号： H04N1/32192 , G06T1/0028 , G06T2201/0051 , G06T2201/0052 , G06T2201/0061 , G10L19/018 , G11B20/00891 , H04N1/32187 , H04N21/23892 , H04N21/8358
摘要： The aim of the invention is to embed binary user data (122) in a carrier signal (110), e.g. an audio signal. Said aim is achieved by transforming a sequence of time-discrete values of the carrier signal into the frequency domain (116) by means of an integer transformation algorithm so as to obtain binary representative spectral values. Bits of the binary representative spectral values, which have a value that is smaller than the signal threshold value, are determined and adjusted according to the user data (120). The signal threshold value for a representative spectral value is smaller than the value of the leading bit of said representative spectral value, resulting in psychoacoustically transparent insertion of data if the distance is sufficiently great such that a modified spectral representation with inserted data is generated which is ultimately transformed back into the time domain by using an integer retransformation algorithm. In order to extract the user data, the time-discrete signal comprising the inserted data is once again transformed into a spectral representation by means of the forward integer transformation algorithm. Moreover, signal threshold value data is determined so as to identify the bits of the binary representative spectral values containing no data regarding the carrier signal but data regarding the user signal in order to extract said bits. The inventive concept is easy to implement and can be scaled regarding the rate of the data that is to be inserted.

76. 发明公开

EP4362014A1 AUDIO SIGNAL ENCODER, AUDIO SIGNAL DECODER, METHOD FOR ENCODING OR DECODING AN AUDIO SIGNAL USING AN ALIASING-CANCELLATION 审中-公开
公开(公告)号：EP4362014A1
公开(公告)日：2024-05-01
申请号：EP24160719.1
申请日：2010-10-19
申请人： Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. , VoiceAge Corporation , Koninklijke Philips N.V. , Dolby International AB
发明人： BESSETTE, Bruno , NEUENDORF, Max , GEIGER, Ralf , GOURNAY, Philippe , LEFEBVRE, Roch , GRILL, Bernhard , LECOMTE, Jérémie , BAYER, Stefan , RETTELBACH, Nikolaus , VILLEMOES, Lars , SALAMI, Redwan , DEN BRINKER, Albertus C.
IPC分类号： G10L19/18 , G10L19/02 , G10L19/04 , G10L19/03 , G10L19/12 , G10L19/20
CPC分类号： G10L19/0212 , G10L19/03 , G10L19/20 , G10L19/18 , G10L19/04 , G10L19/12
摘要： An audio signal decoder (200) for providing a decoded representation (212) of an audio content on the basis of an encoded representation (310) of the audio content comprises a transform domain path (230, 240, 242, 250, 260) configured to obtain a time-domain representation (212) of a portion of the audio content encoded in a transform-domain mode on the basis of a first set (220) of spectral coefficients, a representation (224) of an aliasing-cancellation stimulus signal and a plurality of linear-prediction-domain parameters (222). The transform domain path comprises a spectrum processor (230) configured to apply a spectrum shaping to the first set of spectral coefficients in dependence on at least a subset of the linear-prediction-domain parameters, to obtain a spectrally-shaped version (232) of the first set of spectral coefficients. The transform domain path comprises a first frequency-domain-to-time-domain converter (240) configured to obtain 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 comprises an aliasing-cancellation stimulus filter configured to filter (250) the aliasing-cancellation stimulus signal (324) in dependence on at least a subset of the linear-prediction-domain parameters (222), to derive an aliasing-cancellation synthesis signal (252) from the aliasing-cancellation stimulus signal. The transform domain path also comprises a combiner (260) configured to combine the time-domain representation (242) of the audio content with the aliasing-cancellation synthesis signal (252), or a post-processed version thereof, to obtain an aliasing reduced time-domain signal.

77. 发明公开

EP4246512A3 APPARATUS AND METHOD FOR DECODING OR ENCODING AN AUDIO SIGNAL USING ENERGY INFORMATION VALUES FOR A RECONSTRUCTION BAND 审中-实审
公开(公告)号：EP4246512A3
公开(公告)日：2023-12-13
申请号：EP23188679.7
申请日：2014-07-15
申请人： Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
发明人： NIEDERMEIER, Andreas , ERTEL, Christian , GEIGER, Ralf , GHIDO, Florin , HELMRICH, Christian
IPC分类号： G10L21/0388 , G10L19/02
摘要： An apparatus for decoding an encoded audio signal comprising an encoded representation of a first set of first spectral portions and an encoded representation of parametric data indicating spectral energies for a second set of second spectral portions, comprises: an audio decoder (900) for decoding the encoded representation (901b) of the first set of the first spectral portions to obtain a first set of first spectral portions (904) and for decoding the encoded representation of the parametric data to obtain a decoded parametric data (902) for the second set of second spectral portions indicating, for individual reconstruction bands, individual energies; a frequency regenerator (906) for reconstructing spectral values in a reconstruction band (920) comprising a second spectral portion (922, 923) using a first spectral portion of the first set of the first spectral portions and an individual energy for the reconstruction band, the reconstruction band comprising a first spectral portion (921) and the second spectral portion; wherein the frequency regenerator (906) is configured for determining (912) a survive energy information comprising an accumulated energy of the first spectral portion having frequency values in the reconstruction band, determining (918) a tile energy information of further spectral portions (922, 923) of the reconstruction band (920) for frequency values different from the first spectral portion (921) having frequencies in the reconstruction band (920), wherein the further spectral portions (922, 923) are to be generated by frequency regeneration using a first spectral portion (302) different from the first spectral portion (921, 306) in the reconstruction band; determining (914) a missing energy in the reconstruction band (920) using the individual energy for the reconstruction band and the survive energy information; and adjusting (916) the further spectral portions in the reconstruction band based on the missing energy information and the tile energy information.

78. 发明公开

EP3910628A1 AUDIO ENCODER FOR ENCODING A MULTICHANNEL SIGNAL AND AUDIO DECODER FOR DECODING AN ENCODED AUDIO SIGNAL 有权转让
公开(公告)号：EP3910628A1
公开(公告)日：2021-11-17
申请号：EP21171835.8
申请日：2016-03-07
申请人： FRAUNHOFER-GESELLSCHAFT zur Förderung der angewandten Forschung e.V.
发明人： DISCH, Sascha , FUCHS, Guillaume , RAVELLI, Emmanuel , NEUKAM, Christian , SCHMIDT, Konstantin , BENNDORF, Conrad , NIEDERMEIER, Andreas , SCHUBERT, Benjamin , GEIGER, Ralf
IPC分类号： G10L19/008 , G10L19/02 , G10L19/04 , G10L19/18 , G10L21/038
摘要： A schematic block diagram of an audio encoder 2 for encoding a multichannel audio signal 4 is shown. The audio encoder comprises a linear prediction domain encoder 6, a frequency domain encoder 8, and a controller 10 for switching between the linear prediction domain encoder 6 and the frequency domain encoder 8. The controller is configured such that a portion of the multichannel signal is represented either by an encoded frame of the linear prediction domain encoder or by an encoded frame of the frequency domain encoder. The linear prediction domain encoder comprises a downmixer 12 for downmixing the multichannel signal 4 to obtain a downmixed signal 14. The linear prediction domain encoder further comprises a linear prediction domain core encoder 16 for encoding the downmix signal and furthermore, the linear prediction domain encoder comprises a first joint multichannel encoder 18 for generating first multichannel information 20 from the multichannel signal 4.

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