公开(公告)号：EP2351025B1

公开(公告)日：2012-11-14

申请号：EP10712944.7

申请日：2010-04-01

申请人： Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.

发明人： NAGEL, Frederik ,  MULTRUS, Markus ,  LECOMTE, Jérémie ,  BAYER, Stefan ,  FUCHS, Guillaume ,  HILPERT, Johannes ,  ROBILLIARD, Julien

IPC分类号： G10L21/02 ,  G10L19/14

CPC分类号： G10L19/008 ,  G10L19/18 ,  G10L21/038

公开(公告)号：EP2351025A1

公开(公告)日：2011-08-03

申请号：EP10712944.7

申请日：2010-04-01

申请人： Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.

发明人： NAGEL, Frederik ,  MULTRUS, Markus ,  LECOMTE, Jérémie ,  BAYER, Stefan ,  FUCHS, Guillaume ,  HILPERT, Johannes ,  ROBILLIARD, Julien

IPC分类号： G10L21/02 ,  G10L19/14

CPC分类号： G10L19/008 ,  G10L19/18 ,  G10L21/038

摘要： An apparatus for generating a synthesis audio signal using a patching control signal comprises 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 comprising 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. The high frequency reconstruction manipulator is configured for manipulating the modified spectral representation or a signal derived from the modified spectral representation in accordance with a spectral band replication parameter to obtain a bandwidth extended signal. Finally, the combiner is configured for combining the audio signal having spectral components in the core frequency band or a signal derived from the audio signal with the bandwidth extended signal to obtain the synthesis audio signal.

公开(公告)号：EP2311034A1

公开(公告)日：2011-04-20

申请号：EP09777044.0

申请日：2009-07-08

申请人： Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. ,  VoiceAge Corporation

发明人： LECOMTE, Jérémie ,  GOURNAY, Philippe ,  BAYER, Stefan ,  MULTRUS, Markus ,  RETTELBACH, Nikolaus

IPC分类号： G10L19/14

CPC分类号： G10L19/20

摘要： An audio encoder (100) adapted for encoding frames of a sampled audio signal to obtain encoded frames, wherein a frame comprises a number of time domain audio samples, comprising a predictive coding analysis stage (110) for determining information on coefficients of a synthesis filter and information on a prediction domain frame based on a frame of audio samples. The audio encoder (100) further comprises a frequency domain transformer (120) for transforming a frame of audio samples to the frequency domain to obtain a frame spectrum and an encoding domain decider (130). Moreover, the audio encoder (100) comprises a controller (140) for determining an information on a switching coefficient when the encoding domain decider decides that encoded data of a current frame is based on the information on the coefficients and the information on the prediction domain frame when encoded data of a previous frame was encoded based on a previous frame spectrum.

公开(公告)号：EP4375993A2

公开(公告)日：2024-05-29

申请号：EP24167537.0

申请日：2014-06-16

申请人： Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.

发明人： LECOMTE, Jérémie ,  SCHNABEL, Michael ,  MARKOVIC, Goran ,  DIETZ, Martin ,  NEUGEBAUER, Bernhard

IPC分类号： G10L19/005 ,  G10L25/90 ,  G10L19/107 ,  G10L19/08

CPC分类号： G10L19/107 ,  G10L19/08 ,  G10L25/90 ,  G10L19/005

摘要： An apparatus for determining an estimated pitch lag is provided. The apparatus comprises an input interface (110) for receiving a plurality of original pitch lag values, and a pitch lag estimator (120) for estimating the estimated pitch lag. The pitch lag estimator (120) is configured to estimate the estimated pitch lag depending on a plurality of original pitch lag values and depending on a plurality of information values, wherein for each original pitch lag value of the plurality of original pitch lag values, an information value of the plurality of information values is assigned to said original pitch lag value.

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.

公开(公告)号：EP3427256B1

公开(公告)日：2020-04-08

申请号：EP16725134.7

申请日：2016-05-25

申请人： FRAUNHOFER-GESELLSCHAFT zur Förderung der angewandten Forschung e.V.

发明人： LECOMTE, Jérémie ,  TOMASEK, Adrian

IPC分类号： G10L19/005 ,  G10L19/04 ,  G10L19/02

公开(公告)号：EP3120347B1

公开(公告)日：2018-01-31

申请号：EP15707397.4

申请日：2015-03-04

申请人： Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.

发明人： SCHNABEL, Michael ,  LECOMTE, Jérémie ,  SPERSCHNEIDER, Ralph ,  JANDER, Manuel

IPC分类号： G10L19/005 ,  G10L19/06

CPC分类号： G10L19/005 ,  G10L19/028 ,  G10L19/06 ,  G10L2019/0002

摘要： An apparatus for generating an error concealment signal, comprises: an LPC representation generator (100) for generating a replacement LPC representation; an LPC synthesizer (106, 108) for filtering a codebook information using the replacement LPC representation; and a noise estimator (206) for estimating a noise estimate during a reception of good audio frames, wherein the noise estimate depends on the good audio frames representation generator (100) is configured to use the noise estimate estimated by the noise estimator (206) in generating the replacement LPC representation.

公开(公告)号：EP3011564B1

公开(公告)日：2018-01-31

申请号：EP14733122.7

申请日：2014-06-18

申请人： Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.

发明人： REUSCHL, Stefan ,  DÖHLA, Stefan ,  LECOMTE, Jérémie ,  JANDER, Manuel ,  FÄRBER, Nikolaus

IPC分类号： G10L21/04

CPC分类号： G10L21/04 ,  G10L19/022 ,  G10L25/06

摘要： A time scaler for providing a time scaled version of an input audio signal is configured to compute or estimate a quality of a time scaled version of the input audio signal obtainable by a time scaling of the input audio signal. The time scaler is configured to perform the time scaling of the input audio signal in dependence on the computation or estimation of the quality of the time scaled version of the input audio signal obtainable by the time scaling. An audio decoder has such a time scaler.

公开(公告)号：EP3063760B1

公开(公告)日：2017-12-13

申请号：EP14790073.2

申请日：2014-10-27

申请人： Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.

发明人： LECOMTE, Jérémie ,  MARKOVIC, Goran ,  SCHNABEL, Michael ,  PIETRZYK, Grzegorz

IPC分类号： G10L19/005 ,  G10L25/90 ,  G10L19/08 ,  G10L19/02

CPC分类号： G10L19/005 ,  G10L19/02 ,  G10L19/08 ,  G10L19/09 ,  G10L19/125 ,  G10L2019/0011

公开(公告)号：EP3011558B1

公开(公告)日：2017-07-26

申请号：EP14732194.7

申请日：2014-06-23

申请人： Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.

发明人： SCHNABEL, Michael ,  MARKOVIC, Goran ,  SPERSCHNEIDER, Ralph ,  LECOMTE, Jérémie ,  HELMRICH, Christian

IPC分类号： G10L19/005 ,  G10L25/90

CPC分类号： G10L19/005 ,  G10L19/002 ,  G10L19/012 ,  G10L19/0212 ,  G10L19/06 ,  G10L19/07 ,  G10L19/083 ,  G10L19/09 ,  G10L19/12 ,  G10L19/22 ,  G10L2019/0002 ,  G10L2019/0011 ,  G10L2019/0016

摘要： An apparatus for decoding an encoded audio signal to obtain a reconstructed audio signal is provided, having: a receiving interface for receiving one or more frames, a coefficient generator, and a signal reconstructor. The coefficient generator is configured to determine one or more first audio signal coefficients, and one or more noise coefficients. Moreover, the coefficient generator is configured to generate one or more second audio signal coefficients, depending on the one or more first audio signal coefficients and depending on the one or more noise coefficients. The audio signal reconstructor is configured to reconstruct a first portion of the reconstructed audio signal depending on the one or more first audio signal coefficients and the audio signal reconstructor is configured to reconstruct a second portion of the reconstructed audio signal depending on the one or more second audio signal coefficients, if the current frame is not received by the receiving interface or if the current frame being received by the receiving interface is corrupted.

摘要翻译： 提供了一种用于对编码的音频信号进行解码以获得重建的音频信号的装置，具有：用于接收一个或多个帧的接收接口，系数发生器和信号重构器。 系数发生器被配置为确定一个或多个第一音频信号系数以及一个或多个噪声系数。 此外，系数发生器被配置为根据一个或多个第一音频信号系数并且取决于一个或多个噪声系数来生成一个或多个第二音频信号系数。 音频信号重构器被配置为根据一个或多个第一音频信号系数来重建重建的音频信号的第一部分，并且音频信号重构器被配置为根据一个或多个第二音频信号重建重建的音频信号的第二部分 音频信号系数，如果接收接口没有接收到当前帧，或者接收接口接收到的当前帧被破坏。