专利快速检索-快速检索全球专利，免费商用专利数据库-IPRDB

21. 发明申请

WO2011047886A1 APPARATUS AND METHOD FOR GENERATING A HIGH FREQUENCY AUDIO SIGNAL USING ADAPTIVE OVERSAMPLING 审中-公开
标题翻译：使用自适应超声波产生高频音频信号的装置和方法
公开(公告)号：WO2011047886A1
公开(公告)日：2011-04-28
申请号：PCT/EP2010/057130
申请日：2010-05-25
申请人： DOLBY INTERNATIONAL AB , FRAUNHOFER-GESELLSCHAFT ZUR FÖRDERUNG DER ANGEWANDTEN FORSCHUNG E.V. , VILLEMOES, Lars , EKSTRAND, Per , DISCH, Sascha , NAGEL, Frederik , WILDE, Stefan
发明人： VILLEMOES, Lars , EKSTRAND, Per , DISCH, Sascha , NAGEL, Frederik , WILDE, Stefan
IPC分类号： G10L21/02 , G10L19/02
CPC分类号： G10L21/038 , G10L19/025
摘要： An apparatus for generating a high frequency audio signal that comprises an analyzer (12) for analyzing an input signal to determine a transient information adaptively. Additionally a spectral converter (14) is provided for converting the input signal into an input spectral representation. A spectral processor (13) processes the input spectral representation to generate a processed spectral representation comprising values for higher frequencies than the input spectral representation. A time converter (17) is configured for converting the processed spectral representation to a time representation, wherein the spectral converter or the time converter are controllable to perform a frequency domain oversampling for the first portion of the input signal having the transient information associated and to not perform the frequency domain oversampling for the second portion of the input signal not having the associated transient information.
摘要翻译：一种用于产生高频音频信号的装置，包括分析器（12），用于分析输入信号以自适应地确定瞬时信息。另外，提供频谱转换器（14）用于将输入信号转换成输入频谱表示。频谱处理器（13）处理输入频谱表示以产生包括比输入频谱表示更高频率的值的经处理频谱表示。时间转换器（17）被配置为将经处理的频谱表示转换为时间表示，其中频谱转换器或时间转换器是可控制的，以对具有相关瞬态信息的输入信号的第一部分进行频域过采样，不对不具有相关联的瞬时信息的输入信号的第二部分执行频域过采样。

22. 发明申请

WO2011048117A1 AUDIO SIGNAL ENCODER, AUDIO SIGNAL DECODER, METHOD FOR ENCODING OR DECODING AN AUDIO SIGNAL USING AN ALIASING-CANCELLATION 审中-公开
标题翻译：音频信号编码器，音频信号解码器，使用消息取消来编码或解码音频信号的方法
公开(公告)号：WO2011048117A1
公开(公告)日：2011-04-28
申请号：PCT/EP2010/065752
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 , BESSETTE, Bruno , NEUENDORF, Max , GEIGER, Ralf , GOURNAY, Philippe , LEFEBVRE, Roch , GRILL, Bernhard , LECOMTE, Jeremie , BAYER, Stefan , RETTELBACH, Nikolaus , VILLEMOES, Lars , SALAMI, Redwan , BRINKER, Albertus C. den
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 , BRINKER, Albertus C. den
IPC分类号： G10L19/14 , G10L19/02
CPC分类号： G10L19/03 , G10L19/0212 , G10L19/20 , G10L2019/0008
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.
摘要翻译：用于基于音频内容的编码表示（310）提供音频内容的解码表示（212）的音频信号解码器（200）包括：配置的变换域路径（230,240,242,250,260）基于频谱系数的第一组（220）获得以变换域模式编码的音频内容的一部分的时域表示（212），混叠取消激励信号的表示（224）和多个线性预测域参数（222）。变换域路径包括频谱处理器（230），其被配置为根据线性预测域参数的至少一个子集对第一组频谱系数应用频谱整形，以获得频谱形状的版本（232）的第一组光谱系数。变换域路径包括被配置为基于第一组频谱系数的频谱形式的版本来获得音频内容的时域表示的第一频域 - 时域转换器（240）。变换域路径包括被设置为根据线性预测域参数（222）的至少一个子集过滤（250）混叠取消激励信号（324）的混叠消除刺激滤波器，以导出混叠消除刺激滤波器消除合成信号（252）。变换域路径还包括被配置为将音频内容的时域表示（242）与混叠消除合成信号（252）组合的组合器（260）或其后处理版本，以获得减少的混叠时域信号。

23. 发明申请

WO2010081892A3 CROSS PRODUCT ENHANCED HARMONIC TRANSPOSITION 审中-公开
公开(公告)号：WO2010081892A3
公开(公告)日：2010-07-22
申请号：PCT/EP2010/050483
申请日：2010-01-15
申请人： DOLBY INTERNATIONAL AB , VILLEMOES, Lars , HEDELIN, Per
发明人： VILLEMOES, Lars , HEDELIN, Per
IPC分类号： G10L21/02
摘要： The present invention relates to audio coding systems which make use of a harmonic transposition method for high frequency reconstruction (HFR). A system and a method for generating a high frequency component of a signal from a low frequency component of the signal is described. The system comprises an analysis filter bank providing a plurality of analysis subband signals of the low frequency component of the signal. It also comprises a non-linear processing unit to generate a synthesis subband signal with a synthesis frequency by modifying the phase of a first and a second of the plurality of analysis subband signals and by combining the phase-modified analysis subband signals. Finally, it comprises a synthesis filter bank for generating the high frequency component of the signal from the synthesis subband signal.

24. 发明申请

WO2012000882A1 SELECTIVE BASS POST FILTER 审中-公开
标题翻译：选择性过滤器
公开(公告)号：WO2012000882A1
公开(公告)日：2012-01-05
申请号：PCT/EP2011/060555
申请日：2011-06-23
申请人： DOLBY INTERNATIONAL AB , RESCH, Barbara , KJÖRLING, Kristofer , VILLEMOES, Lars
发明人： RESCH, Barbara , KJÖRLING, Kristofer , VILLEMOES, Lars
IPC分类号： G10L19/14
CPC分类号： G10L19/26 , G10L19/02 , G10L19/0212 , G10L19/032 , G10L19/09 , G10L19/107 , G10L19/12 , G10L19/125 , G10L19/20 , G10L19/22 , G10L19/265 , G10L21/003 , G10L21/007 , G10L21/013
摘要： In one aspect, the invention provides an audio encoding method characterized by a decision being made as to whether the device which will decode the resulting bit stream should apply post filtering including attenuation of interharmonic noise. Hence, the decision whether to use the post filter, which is encoded in the bit stream, is taken separately from the decision as to the most suitable coding mode. In another aspect, there is provided an audio decoding method with a decoding step followed by a post-filtering step, including interharmonic noise attenuation, and being characterized in a step of disabling the post filter in accordance with post filtering information encoded in the bit stream signal. Such a method is well suited for mixed-origin audio signals by virtue of its capability to deactivate the post filter in dependence of the post filtering information only, hence independently of factors such as the current coding mode.
摘要翻译：在一个方面，本发明提供了一种音频编码方法，其特征在于作出关于将对所得比特流进行解码的装置是否应该应用包括衰减间谐波噪声的后置滤波的决定。因此，决定是否使用在比特流中编码的后置滤波器，与最适合的编码模式的决定分开。另一方面，提供了一种音频解码方法，具有解码步骤，随后是包括间谐波噪声衰减的后置滤波步骤，并且其特征在于根据在比特流中编码的后滤波信息来禁用后置滤波器信号。这种方法由于仅根据后滤波信息去激活后置滤波器的能力，因此独立于诸如当前编码模式的因素，这种方法非常适用于混合起始音频信号。

25. 发明申请

WO2011110591A1 AUDIO SIGNAL DECODER, AUDIO SIGNAL ENCODER, METHODS AND COMPUTER PROGRAM USING A SAMPLING RATE DEPENDENT TIME-WARP CONTOUR ENCODING 审中-公开
标题翻译：音频信号解码器，音频信号编码器，使用采样速率的方法和计算机程序依赖于时间波形编码
公开(公告)号：WO2011110591A1
公开(公告)日：2011-09-15
申请号：PCT/EP2011/053538
申请日：2011-03-09
申请人： FRAUNHOFER-GESELLSCHAFT ZUR FÖRDERUNG DER ANGEWANDTEN FORSCHUNG E.V. , DOLBY INTERNATIONAL AB , BAYER, Stefan , BÄCKSTRÖM, Tom , GEIGER, Ralf , EDLER, Bernd , DISCH, Sascha , VILLEMOES, Lars
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 comprising a sampling frequency information, an encoded time warp information and an encoded spectrum representation comprises 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.
摘要翻译：音频信号解码器被配置为基于包括采样频率信息，编码时间扭曲信息和编码频谱表示的编码音频信号表示来提供解码音频信号表示，包括时间扭曲计算器和扭曲解码器。时间扭曲计算器被配置为根据编码时间扭曲信息的码字将映射规则适应于根据采样频率信息描述解码的时间扭曲信息的解码时间扭曲值。翘曲解码器被配置为基于编码的频谱表示并且依赖于解码的时间扭曲信息来提供解码的音频信号表示。

26. 发明申请

WO2011110500A1 APPARATUS AND METHOD FOR PROCESSING AN INPUT AUDIO SIGNAL USING CASCADED FILTERBANKS 审中-公开
标题翻译：使用CASCAD FILTERBANK处理输入音频信号的装置和方法
公开(公告)号：WO2011110500A1
公开(公告)日：2011-09-15
申请号：PCT/EP2011/053315
申请日：2011-03-04
申请人： FRAUNHOFER-GESELLSCHAFT ZUR FÖRDERUNG DER ANGEWANDTEN FORSCHUNG E.V. , DOLBY INTERNATIONAL AB , VILLEMOES, Lars , EKSTRAND, Per , DISCH, Sascha , NAGEL, Frederik , WILDE, Stephan
发明人： VILLEMOES, Lars , EKSTRAND, Per , DISCH, Sascha , NAGEL, Frederik , WILDE, Stephan
IPC分类号： G10L19/02 , G10L21/02 , G10L21/04
CPC分类号： G10L19/008 , G10L19/0204 , G10L21/038 , G10L21/04
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).
摘要翻译：用于处理输入音频信号（2300）的装置依赖于级联的滤波器组，级联包括用于从输入音频信号（2300）合成音频中间信号（2306）的合成滤波器组（2304），输入音频信号为由分析滤波器组（2302）生成的多个第一子带信号（2303）表示，其中合成滤波器组（2304）的多个滤波器组信道小于分析滤波器组（2302）的信道数。该装置还包括用于从音频中间信号（2306）产生多个第二子带信号（2308）的另一分析滤波器组（2307），其中所述另外的分析滤波器组具有与所述音频中间信号合成滤波器组（2304），使得多个第二子带信号（2308）的子带信号的采样率与多个第一子带信号（2303）中的第一子带信号的采样率不同。

27. 发明申请

WO2007051548A1 TIME WARPED MODIFIED TRANSFORM CODING OF AUDIO SIGNALS 审中-公开
标题翻译：时间修改音频信号的变换编码
公开(公告)号：WO2007051548A1
公开(公告)日：2007-05-10
申请号：PCT/EP2006/010246
申请日：2006-10-24
申请人： CODING TECHNOLOGIES AB , VILLEMOES, Lars
发明人： VILLEMOES, Lars
IPC分类号： G10L19/02
CPC分类号： G10L19/002 , G10L19/0212 , G10L19/022
摘要： A spectral representation of an audio signal having consecutive audio frames can be derived more efficiently, when a common time warp is estimated for any two neighbouring frames, such that a following block transform can additionally use the warp information. Thus, window functions required for successful application of an overlap and add procedure during reconstruction can be derived and applied, the window functions already anticipating the re-sampling of the signal due to the time warping. Therefore, the increased efficiency of block-based transform coding of time-warped signals can be used without introducing audible discontinuities.
摘要翻译：当为任何两个相邻帧估计公共时间扭曲时，可以更有效地导出具有连续音频帧的音频信号的频谱表示，使得随后的块变换可以另外使用经线信息。因此，可以导出并应用在重建期间成功应用重叠和添加过程所需的窗口功能，窗函数已经预期由于时间扭曲而对信号的重新采样。因此，可以使用时间扭曲信号的基于块的变换编码的增加的效率而不引入可听不连续性。

28. 发明申请

WO2006108573A1 ADAPTIVE RESIDUAL AUDIO CODING 审中-公开
标题翻译：自适应残留音频编码
公开(公告)号：WO2006108573A1
公开(公告)日：2006-10-19
申请号：PCT/EP2006/003200
申请日：2006-04-07
申请人： CODING TECHNOLOGIES AB , KONINKLIJKE PHILIPS ELECTRONICS N.V. , VILLEMOES, Lars , MYBURG, Francois Philippus
发明人： VILLEMOES, Lars , MYBURG, Francois Philippus
IPC分类号： G10L19/00
CPC分类号： G10L19/008
摘要： An audio signal having at least two channels can be efficiently down-mixed into a downmix signal and a residual signal, when the down-mixing rule used depends on a spatial parameter that is derived from the audio signal and that is post-processed by a limiter to apply a certain limit to the derived spatial parameter with the aim of avoiding instabilities during the up-mixing or down-mixing process. By having a down-mixing rule that dynamically depends on parameters describing an interrelation between the audio channels, one can assure that the energy within the down-mixed residual signal is as minimal as possible, which is advantageous in the view of coding efficiency. By post processing the spatial parameter with a limiter prior to using it in the down-mixing, one can avoid instabilities in the down- or up-mixing, which otherwise could result in a disturbance of the spatial perception of the encoded or decoded audio signal.
摘要翻译：具有至少两个通道的音频信号可以被有效地下混合成降混信号和残留信号，当所使用的下混合规则取决于从音频信号导出的空间参数，并且被后处理时限制器对推导的空间参数应用一定的限制，目的是避免在上混合或下混合过程中的不稳定性。通过具有动态地取决于描述音频通道之间的相互关系的参数的下混合规则，可以确保下混合残差信号内的能量尽可能地最小，这在编码效率方面是有利的。通过在下混合中使用限幅器之前对空间参数进行后处理，可以避免下混合或上混合中的不稳定性，否则可能导致编码或解码的音频信号的空间感知的干扰。

29. 发明申请

WO2004097794A3 ADVANCED PROCESSING BASED ON A COMPLEX-EXPONENTIAL-MODULATED FILTERBANK AND ADAPTIVE TIME SIGNALLING METHODS 审中-公开
公开(公告)号：WO2004097794A3
公开(公告)日：2004-11-11
申请号：PCT/EP2004/004607
申请日：2004-04-30
申请人： CODING TECHNOLOGIES AB , ENGDEGARD, Jonas , VILLEMOES, Lars
发明人： ENGDEGARD, Jonas , VILLEMOES, Lars
IPC分类号： H04S3/00
摘要： A synthesizer for generating a decorrelation signal using an input signal is operative on a plurality of subband signals, wherein a subband signal includes a sequence of at least two subband samples, the sequence of the subband samples representing a bandwidth of the subband signal, which is smaller than a bandwidth of the input signal. The synthesizer includes a filter stage (201) for filtering each subband signal using a reverberation filter to obtain a plurality of reverberated subband signals, wherein a plurality of reverberated subband signals together represent the decorrelation signal. This decorrelation signal is used for reconstructing a signal based on a parametrically encoded stereo signal consisting of a mono signal and a coherence measure.

30. 发明申请

WO2004027998A2 METHOD FOR REDUCTION OF ALIASING INTRODUCED BY SPECTRAL ENVELOPE ADJUSTMENT IN REAL-VALUED FILTERBANKS 审中-公开
标题翻译：用于减少真实值滤波器中光谱环境调整引起的干扰的方法
公开(公告)号：WO2004027998A2
公开(公告)日：2004-04-01
申请号：PCT/EP2003/009485
申请日：2003-08-27
申请人： CODING TECHNOLOGIES AB , KJÖRLING, Kristofer , VILLEMOES, Lars
发明人： KJÖRLING, Kristofer , VILLEMOES, Lars
IPC分类号： H04B
CPC分类号： G10L19/0204 , G10L19/00 , G10L19/025 , G10L19/083 , G10L19/12 , G10L19/26 , G10L25/12 , G10L25/18 , G10L25/21 , G10L25/48 , H03H17/0266
摘要： The present invention proposes a new method for improving the performance of a real-valued filterbank based spectral envelope adjuster. By adaptively locking the gain values for adjacent channels dependent on the sign of the channels, as defined in the application, reduced aliasing is achieved. Furthermore, the grouping of the channels during gain-calculation, gives an improved energy estimate of the real valued subband signals in the filterbank.
摘要翻译：本发明提出了一种用于提高基于实数滤波器组的频谱包络调节器的性能的新方法。通过自适应地锁定取决于通道的符号的相邻通道的增益值，如应用中定义的那样，实现了减少的混叠。此外，增益计算期间信道的分组给出了滤波器组中实值子带信号的改进的能量估计。

你已经成功收藏专利！

检索式保存成功!

IPRDB

热门服务

关于我们

友情链接

联系方式