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    AUDIO SIGNAL DECODER, AUDIO SIGNAL ENCODER, METHOD FOR DECODING AN AUDIO SIGNAL, METHOD FOR ENCODING AN AUDIO SIGNAL AND COMPUTER PROGRAM USING A PITCH-DEPENDENT ADAPTATION OF A CODING CONTEXT
    2.
    发明申请
    AUDIO SIGNAL DECODER, AUDIO SIGNAL ENCODER, METHOD FOR DECODING AN AUDIO SIGNAL, METHOD FOR ENCODING AN AUDIO SIGNAL AND COMPUTER PROGRAM USING A PITCH-DEPENDENT ADAPTATION OF A CODING CONTEXT 有权
    音频信号解码器，音频信号编码器，用于解码音频信号的方法，用于编码音频信号的方法和使用与编码语境相关的适应性的计算机程序

    公开(公告)号：US20130117015A1

    公开(公告)日：2013-05-09

    申请号：US13608980

    申请日：2012-09-10

    申请人： Stefan BAYER ,  Tom BAECKSTROEM ,  Ralf GEIGER ,  Bernd EDLER ,  Sascha DISCH ,  Lars VILLEMOES

    发明人： Stefan BAYER ,  Tom BAECKSTROEM ,  Ralf GEIGER ,  Bernd EDLER ,  Sascha DISCH ,  Lars VILLEMOES

    IPC分类号： G10L19/00

    CPC分类号： G10L19/022 ,  G10L19/0212 ,  G10L25/90

    摘要： An audio signal decoder includes a context-based spectral value decoder configured to decode a codeword describing one or more spectral values or at least a portion of a number representation thereof in dependence on a context state. The audio signal decoder also includes a context state determinator configured to determine a current context state in dependence on one or more previously decoded spectral values and a time warping frequency-domain-to-time-domain converter configured to provide a time-warped time-domain representation of a given audio frame on the basis of a set of decoded spectral values provided by the context-based spectral value decoder and in dependence on the time warp information. The context-state determinator is configured to adapt the determination of the context state to a change of a fundamental frequency between subsequent audio frames. An audio signal encoder applies a comparable concept.

    摘要翻译： 音频信号解码器包括基于上下文的频谱值解码器，其被配置为根据上下文状态来解码描述一个或多个频谱值的码字或其数字表示的至少一部分。 音频信号解码器还包括上下文状态确定器，其被配置为根据一个或多个先前解码的频谱值确定当前上下文状态，以及时间扭曲频域 - 时域转换器，被配置为提供时间扭曲的时域 - 基于由基于上下文的频谱值解码器提供的解码频谱值的集合并且依赖于时间扭曲信息，给定音频帧的域表示。 上下文状态确定器被配置为使上下文状态的确定适应随后的音频帧之间的基频的改变。 音频信号编码器应用可比较的概念。

    METHOD AND APPARATUS FOR ENHANCED VIDEO CODING
    3.
    发明申请
    METHOD AND APPARATUS FOR ENHANCED VIDEO CODING 审中-公开
    用于增强视频编码的方法和装置

    公开(公告)号：US20090022220A1

    公开(公告)日：2009-01-22

    申请号：US11911505

    申请日：2006-04-13

    申请人： Yuri Vatis ,  Bernd Edler ,  Ingolf Wassermann ,  Dieu Thanh Nguyen ,  Joern Ostermann

    发明人： Yuri Vatis ,  Bernd Edler ,  Ingolf Wassermann ,  Dieu Thanh Nguyen ,  Joern Ostermann

    IPC分类号： H04N7/12

    CPC分类号： H04N19/59 ,  H04N19/117 ,  H04N19/136 ,  H04N19/174 ,  H04N19/176 ,  H04N19/182 ,  H04N19/187 ,  H04N19/196 ,  H04N19/31 ,  H04N19/33 ,  H04N19/46 ,  H04N19/463 ,  H04N19/51 ,  H04N19/523 ,  H04N19/593 ,  H04N19/61

    摘要： Standard video compression techniques apply motion-compensated prediction combined with transform coding of the prediction error. In the context of prediction with fractional-pel motion vector resolution it was shown, that aliasing components contained in an image signal are limiting the prediction efficiency obtained by motion compensation. In order to consider aliasing, quantization and motion estimation errors, camera noise, etc., we analytically developed a two dimensional (2D) non-separable interpolation filter, which is independently calculated for each frame by minimizing the prediction error energy. For every fractional-pel position to be interpolated, an individual set of 2D filter coefficients is determined. Since transmitting filter coefficients as side information results in an additional bit rate, which is almost constant for different image resolutions and total bit rates, the loss in coding gain increases when total bit rates sink. Therefore, we developed an algorithm, which regards the non-separable two-dimensional filter as a polyphase filter. For each frame, predicting the interpolation filter impulse response through evaluation of the polyphase filter, we only have to encode the prediction error of the filter coefficients.

    摘要翻译： 标准视频压缩技术应用运动补偿预测与预测误差的变换编码相结合。 在分数像素运动矢量分辨率的预测的上下文中，显示了包含在图像信号中的混叠分量限制了通过运动补偿获得的预测效率。 为了考虑混叠，量化和运动估计误差，相机噪声等，我们分析开发了一种二维（2D）不可分离内插滤波器，它通过最小化预测误差能量对每帧进行独立计算。 对于要插值的每个小数像素位置，确定一组二维滤波器系数。 由于发送滤波器系数作为侧信息导致附加比特率，对于不同的图像分辨率和总比特率几乎是一定的，所以当总比特率下降时，编码增益的损失增加。 因此，我们开发了一种将不可分离的二维滤波器作为多相滤波器的算法。 对于每个帧，通过评估多相滤波器来预测内插滤波器脉冲响应，我们只需要对滤波器系数的预测误差进行编码。

    Audio signal decoder, time warp contour data provider, method and computer program
    4.
    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。

    Device and method for processing a real subband signal for reducing aliasing effects
    5.
    发明授权
    Device and method for processing a real subband signal for reducing aliasing effects 有权
    用于处理实际子带信号以减少混叠效应的装置和方法

    公开(公告)号：US08411731B2

    公开(公告)日：2013-04-02

    申请号：US12375728

    申请日：2007-06-26

    申请人： Bernd Edler ,  Harald Popp

    发明人： Bernd Edler ,  Harald Popp

    IPC分类号： H03K5/159

    CPC分类号： H03G3/00 ,  G10L19/0204 ,  H03H17/0266 ,  H03H2218/04

    摘要： In order to process a subband signal of a plurality of real subband signals which are a representation of a real discrete-time signal generated by an analysis filter bank, a weighter for weighting a subband signal by a weighting factor determined for the subband signal is provided to obtain a weighted subband signal. In addition, a correction term is calculated by a correction term determiner, the correction term determiner being implemented to calculate the correction term using at least one other subband signal and using another weighting factor provided for the other subband signal, the two weighting factors differing. The correction term is then combined with the weighted subband signal to obtain a corrected subband signal, resulting in reduced aliasing, even if subband signals are weighted to a different extent.

    摘要翻译： 为了处理作为由分析滤波器组产生的实际离散时间信号的表示的多个实际子带信号的子带信号，提供了对子带信号进行加权以对于子带信号确定的加权因子的加权器 以获得加权子带信号。 此外，校正项由校正项确定器计算，校正项确定器被实现为使用至少一个其他子带信号来计算校正项，并且使用为另一子带信号提供的另一加权因子，两个加权因子不同。 然后将校正项与加权子带信号组合以获得校正的子带信号，导致减少的混叠，即使子带信号被加权到不同程度。

    Frequency-domain scalable coding without upsampling filters
    8.
    发明授权
    Frequency-domain scalable coding without upsampling filters 有权
    无上采样滤波器的频域可伸缩编码

    公开(公告)号：US06370507B1

    公开(公告)日：2002-04-09

    申请号：US09319066

    申请日：1999-05-28

    申请人： Bernhard Grill ,  Bernd Edler ,  Karlheinz Brandenburg

    发明人： Bernhard Grill ,  Bernd Edler ,  Karlheinz Brandenburg

    IPC分类号： G10L1902

    CPC分类号： G10L19/24 ,  G10L19/0204

    摘要： In a method of coding discrete time signals (X1) sampled with a first sampling rate, second time signals (x2) are generated using the first time signals having a bandwidth corresponding to a second sampling rate, with the second sampling rate being lower than the first sampling rate. The second time signals are coded in accordance with a first coding algorithm. The coded second signals (X2c) are decoded again in order to obtain coded/decoded second time signals (X2cd) having a bandwidth corresponding to the second sampling frequency. The first time signals, by frequency domain transformation, become first spectral values (X1). Second spectral values (X2cd) are generated from the coded/decoded second time signals, the second spectral values being a representation of the coded/decoded time signals in the frequency domain. To obtain weighted spectral values, the first spectral values are weighted by means of the second spectral values, with the first and second spectral values having the same frequency and time resolution. The weighted spectral values (Xb) are coded in accordance with a second coding algorithm in consideration of a psychoacoustic model and written into a bit stream. Weighting the first spectral values and the second spectral values comprises the subtraction of the second spectral values from the first spectral values in to obtain differential spectral values.

    摘要翻译： 在对以第一采样率采样的离散时间信号（X1）进行编码的方法中，使用具有对应于第二采样率的带宽的第一时间信号来生成第二时间信号（x2），其中第二采样率低于 第一次采样率。 第二时间信号根据第一编码算法进行编码。 再次对编码的第二信号（X2c）进行解码，以获得具有对应于第二采样频率的带宽的编码/解码的第二时间信号（X2cd）。 第一次通过频域变换的信号变为第一个光谱值（X1）。 从编码/解码的第二时间信号产生第二频谱值（X2cd），第二频谱值是频域中编码/解码的时间信号的表示。 为了获得加权光谱值，通过第二光谱值对第一光谱值进行加权，其中第一和第二光谱值具有相同的频率和时间分辨率。 考虑到心理声学模型并将其写入比特流，加权频谱值（Xb）根据第二编码算法进行编码。 加权第一光谱值和第二光谱值包括从第一光谱值减去第二光谱值以获得差分光谱值。

    Audio transform coding using pitch correction
    10.
    发明授权
    Audio transform coding using pitch correction 有权
    使用音调校正的音频变换编码

    公开(公告)号：US08700388B2

    公开(公告)日：2014-04-15

    申请号：US12668912

    申请日：2009-03-23

    申请人： Bernd Edler ,  Sascha Disch ,  Ralf Geiger ,  Stefan Bayer ,  Ulrich Kraemer ,  Guillaume Fuchs ,  Max Neuendorf ,  Markus Multrus ,  Gerald Schuller ,  Harald Popp

    发明人： Bernd Edler ,  Sascha Disch ,  Ralf Geiger ,  Stefan Bayer ,  Ulrich Kraemer ,  Guillaume Fuchs ,  Max Neuendorf ,  Markus Multrus ,  Gerald Schuller ,  Harald Popp

    IPC分类号： G10L11/04 ,  G10L21/00 ,  G10L15/00 ,  G10L19/14 ,  G10L19/02 ,  G10L13/06

    CPC分类号： G10L19/0212 ,  G10L19/022

    摘要： A processed representation of an audio signal having a sequence of frames is generated by sampling the audio signal within first and second frames of the sequence of frames, the second frame following the first frame, the sampling using information on a pitch contour of the first and second frames to derive a first sampled representation. The audio signal is sampled within the second and third frames, the third frame following the second frame in the sequence of frames. The sampling uses the information on the pitch contour of the second frame and information on a pitch contour of the third frame to derive a second sampled representation. A first scaling window is derived for the first sampled representation, and a second scaling window is derived for the second sampled representation, the scaling windows depending on the samplings applied to derive the first sampled representations or the second sampled representation.

    摘要翻译： 具有帧序列的音频信号的处理表示是通过对帧序列的第一帧和第二帧中的音频信号进行采样，第一帧之后的第二帧，使用关于第一帧的音调轮廓的信息的采样，以及 第二帧以获得第一采样表示。 音频信号在第二帧和第三帧中采样，第三帧在帧序列中的第二帧之后。 采样使用关于第二帧的音调轮廓的信息和关于第三帧的音调轮廓的信息来导出第二采样表示。 对于第一采样表示导出第一缩放窗口，并且针对第二采样表示导出第二缩放窗口，缩放窗口取决于应用于导出第一采样表示或第二采样表示的采样。