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    Audio signal decoder, time warp contour data provider, method and computer program
    1.
    发明授权
    Audio signal decoder, time warp contour data provider, method and computer program 有权
    音频信号解码器，时间扭曲轮廓数据提供者，方法和计算机程序

    公开(公告)号：US09043216B2

    公开(公告)日：2015-05-26

    申请号：US12935718

    申请日：2009-07-01

    申请人： Stefan Bayer ,  Sascha Disch ,  Ralf Geiger ,  Guillaume Fuchs ,  Max Neuendorf ,  Gerald Schuller ,  Bernd Edler

    发明人： Stefan Bayer ,  Sascha Disch ,  Ralf Geiger ,  Guillaume Fuchs ,  Max Neuendorf ,  Gerald Schuller ,  Bernd Edler

    IPC分类号： G10L21/04 ,  G10L19/00 ,  G10L19/02 ,  G10L19/022

    CPC分类号： G10L21/04 ,  G10L19/0212 ,  G10L19/022 ,  G10L19/032 ,  G10L19/167

    摘要： An audio signal decoder has a time warp contour calculator, a time warp contour data rescaler and a warp decoder. The time warp contour calculator is configured to generate time warp contour data repeatedly restarting from a predetermined time warp contour start value, based on time warp contour evolution information describing a temporal evolution of the time warp contour. The time warp contour data rescaler is configured to rescale at least a portion of the time warp contour data such that a discontinuity at a restart is avoided, reduced or eliminated in a rescaled version of the time warp contour. The warp decoder is configured to provide the decoded audio signal representation, based on an encoded audio signal representation and using the rescaled version of the time warp contour.

    摘要翻译： 音频信号解码器具有时间扭曲轮廓计算器，时间扭曲轮廓数据重定标器和扭曲解码器。 时间扭曲轮廓计算器被配置为基于描述时间扭曲轮廓的时间演变的时间扭曲轮廓演化信息，从预定时间扭曲轮廓开始值产生重复重新起动的时间扭曲轮廓数据。 时间扭曲轮廓数据重定标器被配置为重新缩放时间扭曲轮廓数据的至少一部分，使得在时间扭曲轮廓的重新缩放版本中避免，减少或消除重启时的不连续性。 扭曲解码器被配置为基于编码的音频信号表示并使用时间扭曲轮廓的重新缩放版本来提供解码的音频信号表示。

    Audio encoding/decoding with aliasing switch for domain transforming of adjacent sub-blocks before and subsequent to windowing
    3.
    发明授权
    Audio encoding/decoding with aliasing switch for domain transforming of adjacent sub-blocks before and subsequent to windowing 有权
    使用混叠开关进行音频编码/解码，用于在开窗之前和之后对相邻子块进行域变换

    公开(公告)号：US08862480B2

    公开(公告)日：2014-10-14

    申请号：US13004351

    申请日：2011-01-11

    申请人： Guillaume Fuchs ,  Jeremie Lecomte ,  Stefan Bayer ,  Ralf Geiger ,  Markus Multrus ,  Gerald Schuller ,  Jens Hirschfeld

    发明人： Guillaume Fuchs ,  Jeremie Lecomte ,  Stefan Bayer ,  Ralf Geiger ,  Markus Multrus ,  Gerald Schuller ,  Jens Hirschfeld

    IPC分类号： G10L19/02 ,  G10L19/04 ,  G10L19/20 ,  G10L19/022

    CPC分类号： G10L19/20 ,  G10L19/02 ,  G10L19/022 ,  G10L19/04

    摘要： An apparatus for encoding an audio signal includes the windower for windowing a first block of the audio signal using an analysis window having an aliasing portion and a further portion. The apparatus furthermore includes a processor for processing the first sub-block of the audio signal associated with the aliasing portion by transforming the sub-block from a domain into a different domain subsequent to windowing the first sub-block to obtain the processed first sub-block, and for processing a second sub-block of the audio signal associated with the further portion by transforming the second sub-block from the domain into the different domain before windowing the second sub-block to obtain a processed second sub-block. Thus, a critically sampled switch between two coding modes can be obtained.

    摘要翻译： 用于编码音频信号的装置包括用于使用具有混叠部分和另一部分的分析窗口对音频信号的第一块进行加窗的加窗器。 该装置还包括处理器，用于通过在将第一子块加窗以获得处理的第一子块之后，将子块从域变换为不同的域来处理与混叠部分相关联的音频信号的第一子块， 并且用于通过在开启第二子块之前将第二子块从该域变换为不同的域来处理与另一部分相关联的音频信号的第二子块，以获得经处理的第二子块。 因此，可以获得两种编码模式之间的批量采样开关。

    Audio coder/decoder with predictive coding of synthesis filter and critically-sampled time aliasing of prediction domain frames
    5.
    发明授权
    Audio coder/decoder with predictive coding of synthesis filter and critically-sampled time aliasing of prediction domain frames 有权
    具有合成滤波器的预测编码和预测域帧的临时采样时间混叠的音频编码器/解码器

    公开(公告)号：US08595019B2

    公开(公告)日：2013-11-26

    申请号：US13004475

    申请日：2011-01-11

    申请人： Ralf Geiger ,  Bernhard Grill ,  Bruno Bessette ,  Philippe Gournay ,  Guillaume Fuchs ,  Markus Multrus ,  Max Neuendorf ,  Gerald Schuller

    发明人： Ralf Geiger ,  Bernhard Grill ,  Bruno Bessette ,  Philippe Gournay ,  Guillaume Fuchs ,  Markus Multrus ,  Max Neuendorf ,  Gerald Schuller

    IPC分类号： G10L19/00 ,  G10L19/02 ,  G10L19/04

    CPC分类号： G10L19/0212 ,  G10L19/04

    摘要： An audio encoder adapted for encoding frames of a sampled audio signal to obtain encoded frames, wherein a frame includes a number of time domain audio samples. The audio encoder includes a predictive coding analysis stage for determining information on coefficients of a synthesis filter and a prediction domain frame based on a frame of audio samples. The audio encoder further includes a time-aliasing introducing transformer for transforming overlapping prediction domain frames to the frequency domain to obtain prediction domain frame spectra, wherein the time-aliasing introducing transformer is adapted for transforming the overlapping prediction domain frames in a critically-sampled way. Moreover, the audio encoder includes a redundancy reducing encoder for encoding the prediction domain frame spectra to obtain the encoded frames based on the coefficients and the encoded prediction domain frame spectra.

    摘要翻译： 一种音频编码器，适于编码采样的音频信号的帧以获得编码的帧，其中帧包括多个时域音频采样。 音频编码器包括用于基于音频样本的帧来确定关于合成滤波器和预测域帧的系数的信息的预测编码分析阶段。 音频编码器还包括时间混叠引入变压器，用于将重叠的预测域帧转换到频域以获得预测域帧频谱，其中时间混叠引入变压器适于以重要取样的方式变换重叠的预测域帧 。 此外，音频编码器包括冗余减少编码器，用于根据系数和编码的预测域帧频谱对预测域帧频谱进行编码以获得编码的帧。

    Noise Filler, Noise Filling Parameter Calculator Encoded Audio Signal Representation, Methods and Computer Program
    6.
    发明申请
    Noise Filler, Noise Filling Parameter Calculator Encoded Audio Signal Representation, Methods and Computer Program 有权
    噪声填充，噪声填充参数计算器编码音频信号表示，方法和计算机程序

    公开(公告)号：US20110173012A1

    公开(公告)日：2011-07-14

    申请号：US13004493

    申请日：2011-01-11

    申请人： Nikolaus Rettelbach ,  Bernhard Grill ,  Guillaume Fuchs ,  Stefan Geyersberger ,  Markus Multrus ,  Harald Popp ,  Juergen Herre ,  Stefan Wabnik ,  Gerald Schuller ,  Jens Hirschfeld

    发明人： Nikolaus Rettelbach ,  Bernhard Grill ,  Guillaume Fuchs ,  Stefan Geyersberger ,  Markus Multrus ,  Harald Popp ,  Juergen Herre ,  Stefan Wabnik ,  Gerald Schuller ,  Jens Hirschfeld

    IPC分类号： G10L19/00

    CPC分类号： G10L19/035 ,  G10L19/008 ,  G10L19/02 ,  G10L19/0204 ,  G10L19/028 ,  G10L19/032 ,  G10L25/18

    摘要： A noise filler for providing a noise-filled spectral representation of an audio signal on the basis of an input spectral representation of the audio signal has a spectral region identifier configured to identify spectral regions of the input spectral representation spaced from non-zero spectral regions of the input spectral representation by at least one intermediate spectral region, to obtain identified spectral regions, and a noise inserter configured to selectively introduce noise into the identified spectral regions to obtain the noise-filled spectral representation of the audio signal. A noise filling parameter calculator for providing a noise filling parameter on the basis of a quantized spectral representation of an audio signal has a spectral region identifier, as mentioned above, and a noise value calculator configured to selectively consider quantization errors of the identified spectral regions for a calculation of the noise filling parameter. Accordingly, an encoded audio signal representation representing the audio signal can be obtained.

    摘要翻译： 用于基于音频信号的输入频谱表示来提供音频信号的噪声填充频谱表示的噪声填充器具有频谱区域标识符，其被配置为识别从非零频谱区域间隔开的输入频谱表示的频谱区域 通过至少一个中间光谱区域的输入光谱表示，以获得所识别的光谱区域，以及噪声插入器，被配置为选择性地将噪声引入所识别的光谱区域中，以获得音频信号的噪声填充的频谱表示。 用于根据音频信号的量化频谱表示提供噪声填充参数的噪声填充参数计算器具有如上所述的频谱区域标识符，以及噪声值计算器，被配置为选择性地考虑所识别的频谱区域的量化误差 噪声填充参数的计算。 因此，可以获得表示音频信号的编码音频信号表示。

    Audio coding
    7.
    发明授权
    Audio coding 有权
    音频编码

    公开(公告)号：US07729903B2

    公开(公告)日：2010-06-01

    申请号：US11460425

    申请日：2006-07-27

    申请人： Gerald Schuller ,  Stefan Wabnik ,  Marc Gayer

    发明人： Gerald Schuller ,  Stefan Wabnik ,  Marc Gayer

    IPC分类号： G10L19/00

    CPC分类号： G10L19/032 ,  G10L19/265

    摘要： The central idea of the present invention is that the prior procedure, namely interpolation relative to the filter coefficients and the amplification value, for obtaining interpolated values for the intermediate audio values starting from the nodes has to be dismissed. Coding containing less audible artifacts can be obtained by not interpolating the amplification value, but rather taking the power limit derived from the masking threshold, for each node, i.e. for each parameterization to be transferred, and then performing the interpolation between these power limits of neighboring nodes, such as, for example, a linear interpolation. On both the coder and the decoder side, an amplification value can then be calculated from the intermediate power limit determined such that the quantizing noise caused by quantization, which has a constant frequency before post-filtering on the decoder side, is below the power limit or corresponds thereto after post-filtering.

    摘要翻译： 本发明的中心思想在于，从节点开始的用于从中间音频值获得内插值的先前过程，即相对于滤波器系数和放大值的内插，必须被忽略。 可以通过不内插放大值，而是从每个节点的掩蔽阈值导出的功率极限（即，要传送的每个参数化），然后在相邻的这些功率极限之间进行插值，从而获得包含较少可听见的伪影的编码 节点，例如，线性插值。 在编码器和解码器侧，可以根据所确定的中间功率限制来计算放大值，使得在解码器侧进行后置滤波之前具有恒定频率的量化引起的量化噪声低于功率极限 或者在后处理之后对应于其。

    Device and method for embedding binary payload in a carrier signal
    10.
    发明授权
    Device and method for embedding binary payload in a carrier signal 有权
    用于在载波信号中嵌入二进制有效载荷的装置和方法

    公开(公告)号：US07587311B2

    公开(公告)日：2009-09-08

    申请号：US11274836

    申请日：2005-11-15

    申请人： Gerald Schuller ,  Ralf Geiger ,  Juergen Koller

    发明人： Gerald Schuller ,  Ralf Geiger ,  Juergen Koller

    IPC分类号： 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： For embedding binary payload in a carrier signal, which, for example, is an audio signal, a sequence of time-discrete values of the carrier signal is converted to the frequency domain by means of an integer transform algorithm to obtain binary spectral representation values. Bits of the binary spectral representation values with a valency less than signal limit valency are determined and set according to the payload. The signal limit valency for a spectral representation value is less than the valency of the leading bit of this spectral representation value, so that, with adequate distance, a psychoacoustic transparent insertion of information is achieved. Thus a modified spectral representation with inserted information is generated which is finally converted back to the time domain using an integer back transform algorithm. For extracting the payload, the time-discrete signal with the inserted information is again converted to a spectral representation with the integer forward transform algorithm. Furthermore, signal limit valency information is determined to identify the bits of the binary spectral representation values containing no information regarding the carrier signal, but information regarding the payload signal, to extract these bits. The inventive concept is simple in its implementation and may be scaled with respect to the data rate of the information to be inserted.

    摘要翻译： 为了将二进制有效载荷嵌入到例如音频信号的载波信号中，通过整数变换算法将载波信号的时间离散值的序列转换为频域以获得二进制频谱表示值。 根据有效载荷确定和设置具有小于信号限制价的化合价的二进制光谱表示值的位。 频谱表示值的信号限制价格小于该频谱表示值的前导比特的价格，使得在足够的距离处，可以实现心理声学透明的信息插入。 因此，生成具有插入信息的修改的频谱表示，其最终使用整数反向变换算法转换回时域。 为了提取有效载荷，具有插入信息的时间离散信号再次被转换为具有整数正则变换算法的频谱表示。 此外，确定信号限价信息以识别不包含关于载波信号的信息的二进制频谱表示值的位，但是关于有效负载信号的信息来提取这些位。 本发明的概念在其实现中是简单的，并且可以相对于要插入的信息的数据速率进行缩放。