公开(公告)号：US5579430A

公开(公告)日：1996-11-26

申请号：US380135

申请日：1995-01-26

申请人： Bernhard Grill ,  Karl-Heinz Brandenburg ,  Thomas Sporer ,  Bernd Kurten ,  Ernst Eberlein

发明人： Bernhard Grill ,  Karl-Heinz Brandenburg ,  Thomas Sporer ,  Bernd Kurten ,  Ernst Eberlein

IPC分类号： H03M7/40 ,  H03M7/30 ,  H03M7/42 ,  H04J3/16 ,  G01L3/02 ,  G01L9/00

CPC分类号： H03M7/3053 ,  H03M7/42

摘要： A digital encoding process for transmitting and/or storing acoustical sigs and, in particular, music signals, in which scanned values of the acoustical signal are transformed by means of a transformation or a filter bank into a sequence of second scanned values, which reproduce the spectral composition of the acoustical signal, and the sequence of second scanned values is quantized in accordance with the requirements with varying precision and is partially or entirely encoded by an optimum encoder, and in which a corresponding decoding and inverse transformation takes place during the reproduction. An encoder is utilized in a manner in which the occurrence probability of the quantized spectral coefficient is correlated to the length of the code in such a way that the more frequently the spectral coefficient occurs, the shorter the code word. A code word and, if needed, a supplementary code is allocated to several elements of the sequence or to a value range in order to reduce the size of the table of the encoder. A portion of the code words of variable length are arranged in a raster, and the remaining code words are distributed in the gaps still left so that the beginning of a code word can be more easily found without completely decoding or in the event of faulty transmission.

摘要翻译： 用于发送和/或存储声信号，特别是音乐信号的数字编码过程，其中声信号的扫描值通过变换或滤波器组变换为第二扫描值的序列，其再现 声信号的频谱组成和第二扫描值的序列根据要求具有不同的精度量化，并被最佳编码器部分或全部编码，并且其中在再现期间进行相应的解码和逆变换。 使用编码器，其中量化频谱系数的出现概率与代码的长度相关联，使得频谱系数发生频率越高，码字越短。 如果需要，代码字和补充代码被分配给序列的多个元素或值范围，以便减小编码器的表的大小。 可变长度的码字的一部分被布置在光栅中，并且剩余的码字分布在仍然留下的间隙中，使得可以在没有完全解码的情况下或在发生错误传输的情况下更容易地找到码字的开始 。

公开(公告)号：US08965547B2

公开(公告)日：2015-02-24

申请号：US13593016

申请日：2012-08-23

申请人： Stefan Wabnik ,  Joerg Pickel ,  Bert Greevenbosch ,  Bernhard Grill ,  Ernst Eberlein ,  Giovanni Del Galdo ,  Stefan Kraegeloh ,  Reinhard Zitzmann ,  Tobias Bliem ,  Marco Breiling ,  Juliane Borsum

发明人： Stefan Wabnik ,  Joerg Pickel ,  Bert Greevenbosch ,  Bernhard Grill ,  Ernst Eberlein ,  Giovanni Del Galdo ,  Stefan Kraegeloh ,  Reinhard Zitzmann ,  Tobias Bliem ,  Marco Breiling ,  Juliane Borsum

IPC分类号： G06F17/00 ,  H04K1/02 ,  H04R29/00 ,  G10L19/018 ,  H04H20/14 ,  H04H60/29 ,  H04H60/33

CPC分类号： G10L19/018 ,  H04H20/14 ,  H04H60/29 ,  H04H60/33

摘要： A watermark signal provider provides a watermark signal suitable for being hidden in an audio signal when the watermark signal is added to the audio signal, such that the watermark signal represents watermark data. The watermark signal provider includes a psychoacoustical processor for determining a masking threshold of the audio signal; and a modulator for generating the watermark signal from a superposition of sample-shaping functions spaced apart from each other at a sample time interval of a time-discrete representation of the watermark data, each sample-shaping function being amplitude-weighted with a respective sample of the time-discrete representation, multiplied by a respective amplitude weight depending on the masking threshold, the modulator being configured such that the sample time interval is shorter than a time extension of the sample-shaping functions; and the respective amplitude weight also depends on samples of the time-discrete representation neighboring the respective sample in time.

摘要翻译： 当水印信号被添加到音频信号时，水印信号提供器提供适于隐藏在音频信号中的水印信号，使得水印信号表示水印数据。 水印信号提供器包括用于确定音频信号的掩蔽阈值的心理声学处理器; 以及调制器，用于在水印数据的时间离散表示的采样时间间隔处从彼此间隔开的采样整形功能的叠加产生水印信号，每个采样整形功能用相应样本进行幅度加权 所述时间离散表示根据掩蔽阈值乘以相应的幅度权重，所述调制器被配置为使得所述采样时间间隔短于所述采样整形功能的时间延长; 并且相应的振幅权重也取决于相应样本在时间上相邻的时间离散表示的样本。

公开(公告)号：US08744843B2

公开(公告)日：2014-06-03

申请号：US13449890

申请日：2012-04-18

申请人： Ralf Geiger ,  Guillaume Fuchs ,  Markus Multrus ,  Bernhard Grill

发明人： Ralf Geiger ,  Guillaume Fuchs ,  Markus Multrus ,  Bernhard Grill

IPC分类号： G10L21/00 ,  G10L19/00

CPC分类号： G10L19/12 ,  G10L19/03 ,  G10L19/04 ,  G10L19/083 ,  G10L19/20 ,  G10L2019/0002

摘要： In an embodiment, bitstream elements of sub-frames are encoded differentially to a global gain value so that a change of the global gain value results in an adjustment of an output level of the decoded representation of the audio content. Concurrently, the differential coding saves bits. Even further, the differential coding enables the lowering of the burden of globally adjusting the gain of an encoded bitstream. In another embodiment, a global gain control across CELP coded frames and transform coded frames is achieved by co-controlling the gain of the codebook excitation of the CELP codec, along with a level of the transform or inverse transform of the transform coded frames. In another embodiment, the gain value determination in CELP coding is performed in the weighted domain of the excitation signal.

摘要翻译： 在一个实施例中，子帧的比特流元素被差分地编码为全局增益值，使得全局增益值的改变导致音频内容的解码表示的输出电平的调整。 同时，差分编码保存位。 此外，差分编码能够降低全局调整编码比特流的增益的负担。 在另一个实施例中，跨CELP编码帧和变换编码帧的全局增益控制与CELP编解码器的码本激励的增益以及变换编码帧的变换或逆变换的级别一起共同实现。 在另一实施例中，在激励信号的加权域中执行CELP编码中的增益值确定。

公开(公告)号：US08682652B2

公开(公告)日：2014-03-25

申请号：US12305936

申请日：2007-05-16

申请人： Juergen Herre ,  Bernhard Grill ,  Markus Multrus ,  Stefan Bayer ,  Ulrich Kraemer ,  Jens Hirschfeld ,  Stefan Wabnik ,  Gerald Schuller

发明人： Juergen Herre ,  Bernhard Grill ,  Markus Multrus ,  Stefan Bayer ,  Ulrich Kraemer ,  Jens Hirschfeld ,  Stefan Wabnik ,  Gerald Schuller

IPC分类号： G10L19/00

CPC分类号： G10L19/22 ,  G10L19/022

摘要： An audio encoder, an audio decoder or an audio processor includes a filter for generating a filtered audio signal, the filter having a variable warping characteristic, the characteristic being controllable in response to a time-varying control signal, the control signal indicating a small or no warping characteristic or a comparatively high warping characteristic. Furthermore, a controller is connected for providing the time-varying control signal, which depends on the audio signal. The filtered audio signal can be introduced to an encoding processor having different encoding algorithms, one of which is a coding algorithm adapted to a specific signal pattern. Alternatively, the filter is a post-filter receiving a decoded audio signal.

摘要翻译： 音频编码器，音频解码器或音频处理器包括用于产生经滤波的音频信号的滤波器，该滤波器具有可变的翘曲特性，该特性可响应于时变控制信号而被控制，该控制信号指示小的或 无翘曲特性或较高翘曲特性。 此外，连接控制器以提供取决于音频信号的时变控制信号。 滤波后的音频信号可以被引入到具有不同编码算法的编码处理器中，其中之一是适用于特定信号模式的编码算法。 或者，滤波器是接收经解码的音频信号的后置滤波器。

公开(公告)号：US20120271644A1

公开(公告)日：2012-10-25

申请号：US13449949

申请日：2012-04-18

申请人： Bruno Bessette ,  Max Neuendorf ,  Ralf Geiger ,  Philippe Gournay ,  Roch Lefebvre ,  Bernhard Grill ,  Jeremie Lecomte ,  Stefan Bayer ,  Nikolaus Rettelbach ,  Lars Villemoes ,  Redwan Salami ,  Albertus C. Den Brinker

发明人： Bruno Bessette ,  Max Neuendorf ,  Ralf Geiger ,  Philippe Gournay ,  Roch Lefebvre ,  Bernhard Grill ,  Jeremie Lecomte ,  Stefan Bayer ,  Nikolaus Rettelbach ,  Lars Villemoes ,  Redwan Salami ,  Albertus C. Den Brinker

IPC分类号： G10L19/12

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变换域路径，被配置为基于第一组频谱系数获得音频内容的一部分的时域表示，混叠取消激励信号的表示和多个线性 - 预测域参数。 变换域路径将频谱整形应用于第一组频谱系数，以获得其频谱形状的版本。 变换域路径基于第一组频谱系数的频谱形式获得音频内容的时域表示。 变换域路径包括混叠消除刺激滤波器，以根据线性预测域参数的至少一个子集来滤除混叠消除激励信号。 变换域路径还包括组合器，其被配置为将音频内容的时域表示与混叠消除合成信号组合以获得混叠减小的时域信号。

公开(公告)号：US08275626B2

公开(公告)日：2012-09-25

申请号：US13004272

申请日：2011-01-11

申请人： Max Neuendorf ,  Bernhard Grill ,  Ulrich Kraemer ,  Markus Multrus ,  Harald Popp ,  Nikolaus Rettelbach ,  Frederik Nagel ,  Markus Lohwasser ,  Marc Gayer ,  Manuel Jander ,  Virgilio Bacigalupo

发明人： Max Neuendorf ,  Bernhard Grill ,  Ulrich Kraemer ,  Markus Multrus ,  Harald Popp ,  Nikolaus Rettelbach ,  Frederik Nagel ,  Markus Lohwasser ,  Marc Gayer ,  Manuel Jander ,  Virgilio Bacigalupo

IPC分类号： G10L19/00

CPC分类号： G10L21/038 ,  G10L19/008 ,  G10L19/0212 ,  G10L19/022 ,  G10L19/20 ,  G10L21/04

摘要： An apparatus for decoding an encoded audio signal having first and second portions encoded in accordance with first and second encoding algorithms, respectively, BWE parameters for the first and second portions and a coding mode information indicating a first or a second decoding algorithm, includes first and second decoders, a BWE module and a controller. The decoders decode portions in accordance with decoding algorithms for time portions of the encoded signal to obtain decoded signals. The BWE module has a controllable crossover frequency and is configured for performing a bandwidth extension algorithm using the first decoded signal and the BWE parameters for the first portion, and for performing a bandwidth extension algorithm using the second decoded signal and the bandwidth extension parameter for the second portion. The controller controls the crossover frequency for the BWE module in accordance with the coding mode information.

摘要翻译： 一种用于解码具有根据第一和第二编码算法编码的第一和第二部分的编码音频信号的装置，分别用于第一和第二部分的BWE参数和指示第一或第二解码算法的编码模式信息包括第一和第二编码算法 第二解码器，BWE模块和控制器。 解码器根据编码信号的时间部分的解码算法来解码部分，以获得解码信号。 BWE模块具有可控的交叉频率，并且被配置为使用第一解码信号和第一部分的BWE参数来执行带宽扩展算法，并且用于使用第二解码信号和带宽扩展参数来执行带宽扩展算法 第二部分。 控制器根据编码模式信息控制BWE模块的交叉频率。

公开(公告)号：US08037114B2

公开(公告)日：2011-10-11

申请号：US11762690

申请日：2007-06-13

申请人： Marc Gayer ,  Manfred Lutzky ,  Markus Lohwasser ,  Sascha Disch ,  Johannes Hilpert ,  Stefan Geyersberger ,  Bernhard Grill

发明人： Marc Gayer ,  Manfred Lutzky ,  Markus Lohwasser ,  Sascha Disch ,  Johannes Hilpert ,  Stefan Geyersberger ,  Bernhard Grill

IPC分类号： G06F15/00

CPC分类号： G06F7/544 ,  G06F7/49936 ,  G06F7/49942 ,  G06F7/5443 ,  G10L19/02

摘要： In the transition into the logarithmic range, not the entire bit width of the result linearly dependent upon the square of the value must be considered. Rather, it is possible to scale the result of a value with x bits such that a representation with less than x bits of the result is sufficient to receive the logarithmic representation based thereon. The effect of the scaling factor on the resulting logarithmic representation may be compensated for by adding or subtracting a correction value received by the logarithm function applied to the scaling factor to or from the scaled logarithmic representation without any loss of dynamics. This way, a method and an apparatus for creating a representation of a result linearly dependent upon a square of a value are provided so that the calculation is simple and/or possible with little hardware expenditure.

摘要翻译： 在向对数范围的转换中，必须考虑线性依赖于该值平方的结果的整数位宽度。 相反，可以用x比特来缩放值的结果，使得具有小于x位的结果的表示足以基于此来接收对数表示。 缩放因子对所得到的对数表示的影响可以通过将由应用于比例因子的对数函数接收的校正值加到或从缩放的对数表示中减去而没有任何动态损失来补偿。 这样，提供了一种用于创建线性表示的方法和装置，该结果线性地取决于值的平方，使得在少量硬件消耗的情况下计算是简单的和/或可能的。

公开(公告)号：US08036903B2

公开(公告)日：2011-10-11

申请号：US11744641

申请日：2007-05-04

申请人： Bernhard Grill ,  Markus Schnell ,  Ralf Geiger ,  Gerald Schuller

发明人： Bernhard Grill ,  Markus Schnell ,  Ralf Geiger ,  Gerald Schuller

IPC分类号： G10L19/00 ,  G10L19/14

CPC分类号： G10L19/135 ,  G10L19/0212 ,  G10L19/022

摘要： An embodiment of an analysis filterbank for filtering a plurality of time domain input frames, wherein an input frame comprises a number of ordered input samples, comprises a windower configured to generate a plurality of windowed frames, wherein a windowed frame comprises a plurality of windowed samples, wherein the windower is configured to process the plurality of input frames in an overlapping manner using a sample advance value, wherein the sample advance value is less than the number of ordered input samples of an input frame divided by two, and a time/frequency converter configured to provide an output frame comprising a number of output values, wherein an output frame is a spectral representation of a windowed frame.

摘要翻译： 一种用于对多个时域输入帧进行滤波的分析滤波器组的实施例，其中输入帧包括多个有序输入样本，包括被配置为生成多个窗口化帧的加窗器，其中，窗口化框架包括多个窗口样本 其中，所述加权器被配置为使用样本提前值以重叠的方式处理所述多个输入帧，其中所述样本提前值小于输入帧除以2的有序输入样本的数量，并且时间/频率 转换器，被配置为提供包括多个输出值的输出帧，其中输出帧是窗口化帧的频谱表示。

公开(公告)号：US20110200198A1

公开(公告)日：2011-08-18

申请号：US13004453

申请日：2011-01-11

申请人： Bernhard Grill ,  Stefan Bayer ,  Guillaume Fuchs ,  Stefan Geyersberger ,  Ralf Geiger ,  Johannes Hilpert ,  Ulrich Kraemer ,  Jeremie Lecomte ,  Markus Multrus ,  Max Neuendorf ,  Harald Popp ,  Nikolaus Rettelbach ,  Frederik Nagel ,  Sascha Disch ,  Juergen Herre ,  Yoshikazu Yokotani ,  Stefan Wabnik ,  Gerald Schuller ,  Jens Hirschfeld

发明人： Bernhard Grill ,  Stefan Bayer ,  Guillaume Fuchs ,  Stefan Geyersberger ,  Ralf Geiger ,  Johannes Hilpert ,  Ulrich Kraemer ,  Jeremie Lecomte ,  Markus Multrus ,  Max Neuendorf ,  Harald Popp ,  Nikolaus Rettelbach ,  Frederik Nagel ,  Sascha Disch ,  Juergen Herre ,  Yoshikazu Yokotani ,  Stefan Wabnik ,  Gerald Schuller ,  Jens Hirschfeld

IPC分类号： G10L19/14 ,  G10L21/00

CPC分类号： G10L19/173 ,  G10L19/0017 ,  G10L19/008 ,  G10L19/0212 ,  G10L19/18 ,  G10L2019/0008

摘要： An audio encoder has a common preprocessing stage, an information sink based encoding branch such as spectral domain encoding branch, a information source based encoding branch such as an LPC-domain encoding branch and a switch for switching between these branches at inputs into these branches or outputs of these branches controlled by a decision stage. An audio decoder has a spectral domain decoding branch, an LPC-domain decoding branch, one or more switches for switching between the branches and a common post-processing stage for post-processing a time-domain audio signal for obtaining a post-processed audio signal.

摘要翻译： 音频编码器具有通用的预处理级，诸如频域编码分支的基于信息宿的编码分支，诸如LPC域编码分支的基于信息源的编码分支以及用于在这些分支的输入处切换这些分支之间的切换，或 这些分支的输出由决策阶段控制。 音频解码器具有频域解码分支，LPC域解码分支，用于在分支之间切换的一个或多个开关和用于后处理时域音频信号以用于获得后处理音频的公共后处理级 信号。

公开(公告)号：US20110173007A1

公开(公告)日：2011-07-14

申请号：US13004282

申请日：2011-01-11

申请人： Markus Multrus ,  Bernhard Grill ,  Guillaume Fuchs ,  Stefan Geyersberger ,  Nikolaus Rettelbach ,  Virgilio Bacigalupo

发明人： Markus Multrus ,  Bernhard Grill ,  Guillaume Fuchs ,  Stefan Geyersberger ,  Nikolaus Rettelbach ,  Virgilio Bacigalupo

IPC分类号： G10L19/00

CPC分类号： G10L19/02 ,  G10L19/022 ,  H03M7/30

摘要： An audio encoder for encoding segments of coefficients, the segments of coefficients representing different time or frequency resolutions of a sampled audio signal, the audio encoder including a processor for deriving a coding context for a currently encoded coefficient of a current segment based on a previously encoded coefficient of a previous segment, the previously encoded coefficient representing a different time or frequency resolution than the currently encoded coefficient. The audio encoder further includes an entropy encoder for entropy encoding the current coefficient based on the coding context to obtain an encoded audio stream.

摘要翻译： 一种用于对系数段进行编码的音频编码器，表示采样音频信号的不同时间或频率分辨率的系数段，音频编码器包括用于基于先前编码的当前段的当前编码系数导出编码上下文的处理器 前一段的系数，先前编码的系数表示与当前编码系数不同的时间或频率分辨率。 音频编码器还包括熵编码器，用于基于编码上下文对当前系数进行熵编码，以获得经编码的音频流。