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1. 发明授权

US06750789B2 Device and method for determining a coding block raster of a decoded signal 有权
标题翻译：用于确定解码信号的编码块光栅的装置和方法
公开(公告)号：US06750789B2
公开(公告)日：2004-06-15
申请号：US10168456
申请日：2002-10-25
申请人： Juergen Herre , Karlheinz Brandenburg , Thomas Sporer , Michael Schug , Wolfgang Schildbach
发明人： Juergen Herre , Karlheinz Brandenburg , Thomas Sporer , Michael Schug , Wolfgang Schildbach
IPC分类号： H03M700
CPC分类号： G10L19/02
摘要： In determining a coding block raster on which a decoded signal is based, a segment of the decoded signal is picked out first, said segment beginning at a certain output sampling value of the decoded signal. Said segment is then converted into a spectral representation, whereupon said spectral representation is then evaluated in relation to a predetermined criterion in order to obtain an evaluation result for the segment. This procedure is repeated for a plurality of different segments beginning at different output sampling values each, in order to obtain a plurality of evaluation results. Finally, the plurality of the evaluation results is searched in order to establish the evaluation result that has an extreme value as compared to the other evaluation results, in such a way that it can be assumed that the segment to which this evaluation result is allocated matches the coding block raster on which the decoded signal is based. This method can be used to determine the coding block raster for any decoded signal that has no explicit information about its coding block raster.
摘要翻译：在确定解码信号所基于的编码块光栅时，首先选择解码信号的片段，所述片段以解码信号的特定输出采样值开始。然后将所述段转换为频谱表示，然后根据预定标准对所述频谱表示进行评估，以获得该段的评估结果。对于从不同的输出采样值开始的多个不同的段重复该过程，以获得多个评估结果。最后，搜索多个评估结果以便建立与其他评估结果相比具有极值的评估结果，使得可以假定分配了该评估结果的分段匹配解码信号所基于的编码块光栅。该方法可以用于确定任何没有关于其编码块光栅的显式信息的解码信号的编码块栅格。

2. 发明申请

US20050175252A1 Device and method for analysing a decoded time signal 有权
标题翻译：用于分析解码时间信号的装置和方法
公开(公告)号：US20050175252A1
公开(公告)日：2005-08-11
申请号：US10220651
申请日：2001-02-16
申请人： Juergen Herre , Martin Dietz , Thomas Sporer , Michael Schug , Wolfgang Schildbach
发明人： Juergen Herre , Martin Dietz , Thomas Sporer , Michael Schug , Wolfgang Schildbach
IPC分类号： H04N19/00 , H04N19/126 , H04N19/40 , H04N19/60 , G06K9/36
CPC分类号： H04N19/00 , H04N19/126 , H04N19/40 , H04N19/60
摘要： An apparatus for analyzing an analysis time signal that has been generated from encoding and decoding an original time signal according to an encoding algorithm first, wherein first the encoding block raster underlying the analysis time signal used by the encoding algorithm is determined. Thereupon, the analysis time signal will be converted from its timely representation comprising a plurality of analysis spectral coefficients, to a spectral representation by using the established encoding block raster. Then, at least two analysis spectral coefficients or at least two spectral coefficients derived from the analysis spectral coefficients by multiplication of an encoding amplification factor or by multiplication with a compression function are grouped. Then, the greatest common divisor of the analysis spectral coefficients or the spectral coefficients derived from the analysis spectral coefficients will be calculated, corresponding to the quantization step width used when quantizing the encoding algorithm or an integer multiple of it. Then, in the case of an audio signal, the scale factor can easily be established for this group of spectral coefficients, i.e. for a scale factor band, from the quantization step width. Thus, all parameters used for the quantization of the original time signal are known, so that for quantizing the analysis time signal no longer full iteration loops have to be performed, which are, on the one hand, very computing time intensive and, on the other hand, introduce tandem encoding distortions.
摘要翻译：一种用于分析根据编码算法首先对原始时间信号进行编码和解码而产生的分析时间信号的装置，其中首先确定编码算法使用的分析时间信号下面的编码块光栅。因此，分析时间信号将通过使用所建立的编码块光栅从包括多个分析频谱系数的及时表示转换成频谱表示。然后，将通过编码放大因子的乘法或通过与压缩函数相乘而从分析频谱系数导出的至少两个分析频谱系数或至少两个频谱系数分组。然后，对应于当量化编码算法或其整数倍时使用的量化步长，将计算分析频谱系数的最大公约数或从分析频谱系数导出的频谱系数。然后，在音频信号的情况下，从量化步长可以容易地为该组频谱系数建立比例因子，即缩放因子频带。因此，用于原始时间信号的量化的所有参数是已知的，使得对于分析时间信号的量化不再必须执行完整的迭代循环，这一方面一方面非常计算时间密集，并且在另一方面，引入串联编码失真。

3. 发明授权

US07181079B2 Time signal analysis and derivation of scale factors 有权
标题翻译：时间信号分析和推理的比例因子
公开(公告)号：US07181079B2
公开(公告)日：2007-02-20
申请号：US10220651
申请日：2001-02-16
申请人： Juergen Herre , Martin Dietz , Thomas Sporer , Michael Schug , Wolfgang Schildbach
发明人： Juergen Herre , Martin Dietz , Thomas Sporer , Michael Schug , Wolfgang Schildbach
IPC分类号： G06K9/36 , H04B1/66 , H04N7/12 , H04N11/02 , H04N11/04 , G10L19/00 , G10L21/00
CPC分类号： H04N19/00 , H04N19/126 , H04N19/40 , H04N19/60
摘要： Analyzing an analysis time signal that has been generated from encoding and decoding and original time signal according to an encoding algorithm. The encoding block raster underlying the analysis time signal used by the encoding algorithm is determined. The analysis time signal is converted from its timely representation of analysis spectral coefficients to a spectral representation by using the established encoding block raster. At least two analysis spectral coefficients are grouped. The greatest common divisor of the analysis spectral coefficients are calculated, corresponding to the quantization step width used when quantizing the encoding algorithm or an integer multiple of it. In the case of an audio signal, the scale factor can easily be established for this group of spectral coefficients, i.e., for a scale factor band, from the quantization step width. All parameters used for the quantization of the original time signal are known; full iteration loops need not be performed.
摘要翻译：根据编码算法分析从编码和解码产生的分析时间信号和原始时间信号。确定由编码算法使用的分析时间信号的编码块光栅。分析时间信号通过使用建立的编码块光栅从分析频谱系数的及时表示转换为频谱表示。至少两个分析光谱系数被分组。计算分析频谱系数的最大公约数，对应于量化编码算法时使用的量化步长或其整数倍。在音频信号的情况下，从量化步长可以容易地为该组频谱系数（即，比例因子频带）建立比例因子。用于量化原始时间信号的所有参数是已知的; 不需要执行完整的迭代循环。

4. 发明授权

US06421802B1 Method for masking defects in a stream of audio data 有权
标题翻译：用于掩蔽音频数据流中的缺陷的方法
公开(公告)号：US06421802B1
公开(公告)日：2002-07-16
申请号：US09331697
申请日：1999-06-23
申请人： Wolfgang Schildbach , Jürgen Herre , Martin Sieler , Karlheinz Brandenburg , Heinz Gerhäuser
发明人： Wolfgang Schildbach , Jürgen Herre , Martin Sieler , Karlheinz Brandenburg , Heinz Gerhäuser
IPC分类号： G10L1900
CPC分类号： G11B20/1876 , G10L19/005 , G11B20/00992 , H04H60/11
摘要： In a method for concealing errors in an audio data stream the occurrence of an error is detected in the audio data stream, audio data prior to the occurrence of the fault being intact audio data. Thereafter a spectral energy of a subgroup of the intact audio data is calculated. After forming a pattern for substitute data on the basis of the spectral energy calculated for the subgroup of the intact audio data, substitute data for erroneous or missing audio data which correspond to the subgroup are created on the basis of the pattern.
摘要翻译：在用于隐藏音频数据流中的错误的方法中，在音频数据流中检测到错误的发生，在发生故障之前的音频数据是完整的音频数据。此后，计算完整音频数据的子组的频谱能量。基于为完整音频数据的子组计算的频谱能量形成用于替代数据的模式之后，基于该模式创建与该子组对应的错误或缺失音频数据的替代数据。

5. 发明申请

US20080255856A1 Audio Encoding and Decoding 有权
标题翻译：音频编码和解码
公开(公告)号：US20080255856A1
公开(公告)日：2008-10-16
申请号：US11995538
申请日：2006-07-07
申请人： Erik Gosuinus Petrus Schuijers , Gerard Herman Hotho , Heiko Purnhagen , Wolfgang Schildbach , Holger Horich , Hans Magnus Kristofer Kjorling , Karl Jonas Roden
发明人： Erik Gosuinus Petrus Schuijers , Gerard Herman Hotho , Heiko Purnhagen , Wolfgang Schildbach , Holger Horich , Hans Magnus Kristofer Kjorling , Karl Jonas Roden
IPC分类号： G10L21/00
CPC分类号： G10L19/008 , H04S2420/03
摘要： An audio encoder (109) has a hierarchical encoding structure and generates a data stream comprising one or more audio channels as well as parametric audio encoding data. The encoder (109) comprises an encoding structure processor (305) which inserts decoder tree structure data into the data stream. The decoder tree structure data comprises at least one data value indicative of a channel split characteristic for an audio channel at a hierarchical layer of the hierarchical decoder structure and may specifically specify the decoder tree structures to be applied by a decoder A decoder (115) comprises a receiver (401) which receives the data stream and a decoder structure processor (405) for generating the hierarchical decoder structure in response to the decoder tree structure data. A decode processor (403) then generates output audio channels from the data stream using the hierarchical decoder structure.
摘要翻译：音频编码器（109）具有分层编码结构，并且生成包括一个或多个音频信道以及参数音频编码数据的数据流。编码器（109）包括将解码器树结构数据插入到数据流中的编码结构处理器（305）。解码器树结构数据包括指示分层解码器结构的分级层处的音频信道的信道分离特性的至少一个数据值，并且可以具体指定要由解码器A应用的解码器树结构。解码器（115）包括接收数据流的接收机（401）和用于响应于解码器树结构数据生成分层解码器结构的解码器结构处理器（405）。解码处理器（403）然后使用分层解码器结构从数据流生成输出音频通道。

6. 发明申请

US20130253917A1 PSYCHOACOUSTIC FILTER DESIGN FOR RATIONAL RESAMPLERS 审中-公开
标题翻译： PSYCHOACOUSTIC过滤器设计用于RATIONAL RESAMPLERS
公开(公告)号：US20130253917A1
公开(公告)日：2013-09-26
申请号：US13992344
申请日：2011-12-09
申请人： Wolfgang Schildbach
发明人： Wolfgang Schildbach
IPC分类号： G10L21/0232 , G06T5/00
CPC分类号： G10L21/0232 , G06T5/00 , G10L19/032 , G10L19/173 , G10L19/26 , H03H17/0685
摘要： The present document relates to the design of anti-aliasing and/or anti-imaging filters for resamplers using rational resampling factors. In particular, the present document relates to a method for designing such filters having a reduced number of filter coefficients or an increased perceptual performance, as well as to the filters designed using such method. A method for designing a filter (102) configured to reduce imaging and/or aliasing of an output audio signal (113) at an output sampling rate (fsout) is described. The output audio signal (113) is a resampled version of an input audio signal (110) at an input sampling rate (fsin). The ratio of the output sampling rate (fsout) and the input sampling rate (fsin) is a rational number N/M. The filter (102) operates at an upsampled sampling rate which equals N times the input sampling rate (fsin). The method comprises the steps of selecting an allowed deviation of the frequency response (531, 532) of the filter (102) within a stop band of the filter (102) based on a perceptual frequency response indicative of an auditory spectral sensitivity; wherein the allowed deviation indicates a deviation of the frequency response (531, 532) of the filter (102) from a predetermined attenuation within the stop band; and of determining coefficients of the filter (102) such that the frequency response (531, 532) of the filter (102) is fitted to the allowed deviation of the frequency response (531, 532).
摘要翻译：本文件涉及使用合理的重采样因子的重采样器的抗混叠和/或抗成像滤波器的设计。特别地，本文件涉及一种用于设计具有减少的滤波器系数数量或增加的感知性能的滤波器的方法，以及使用这种方法设计的滤波器。描述了一种用于设计滤波器（102）的方法，其被配置为以输出采样率（fsout）来减少输出音频信号（113）的成像和/或混叠。输出音频信号（113）是输入采样率（fsin）的输入音频信号（110）的再采样版本。输出采样率（fsout）和输入采样率（fsin）的比值是有理数N / M。滤波器（102）以等于输入采样率（fsin）的N倍的上采样率进行操作。该方法包括以下步骤：基于指示听觉光谱灵敏度的感知频率响应，在滤波器（102）的阻带内选择滤波器（102）的频率响应（531,532）的允许偏差; 其中所述允许偏差指示所述滤波器（102）的频率响应（531,532）与所述阻带内的预定衰减的偏差; 以及确定所述滤波器（102）的系数，使得所述滤波器（102）的频率响应（531,532）适应于所述频率响应（531,532）的允许偏差。

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