公开(公告)号：US20110158415A1

公开(公告)日：2011-06-30

申请号：US12935740

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： H04R5/00

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

摘要： An audio signal decoder for providing a decoded multi-channel audio signal representation on the basis of an encoded multi-channel audio signal representation has a time warp decoder configured to selectively use individual audio channel specific time warp contours or a joint multi-channel time warp contour for a reconstruction of a plurality of audio channels represented by the encoded multi-channel audio signal representation. An audio signal encoder for providing an encoded representation of a multi-channel audio signal has an encoded audio representation provider configured to selectively provide an audio representation having a common time warp contour information, commonly associated with a plurality of audio channels of the multi-channel audio signal, or an encoded audio representation having individual time warp contour information, individually associated with the different audio channels of the plurality of audio channels, in dependence on an information describing a similarity or difference between time warp contours associated with the audio channels of the plurality of audio channels.

摘要翻译： 用于基于编码的多声道音频信号表示提供解码的多声道音频信号表示的音频信号解码器具有时间扭曲解码器，其被配置为选择性地使用单独的音频通道特定时间扭曲轮廓或联合多通道时间扭曲 用于重建由编码的多声道音频信号表示表示的多个音频通道的轮廓。 用于提供多声道音频信号的编码表示的音频信号编码器具有编码音频表示提供器，其被配置为选择性地提供具有公共时间扭曲轮廓信息的音频表示，通常与多声道音频信号的多个音频通道相关联 音频信号或具有单独与多个音频通道的不同音频通道相关联的具有各个时间扭曲轮廓信息的经编码的音频表示，该信息根据描述与音频通道的音频通道相关联的时间扭曲轮廓之间的相似性或差异的信息 多个音频通道。

公开(公告)号：US20110106529A1

公开(公告)日：2011-05-05

申请号：US12922823

申请日：2009-03-10

申请人： Sascha Disch

发明人： Sascha Disch

IPC分类号： G10L19/00

CPC分类号： G10L19/0204 ,  G10L19/09 ,  G10L19/16 ,  G10L19/20 ,  G10L25/90

摘要： Apparatus for converting an audio signal into a parameterized representation, has a signal analyzer for analyzing a portion of the audio signal to obtain an analysis result; a band pass estimator for estimating information of a plurality of band pass filters based on the analysis result, wherein the information on the plurality of band pass filters has information on a filter shape for the portion of the audio signal, wherein the band width of a band pass filter is different over an audio spectrum and depends on the center frequency of the band pass filter; a modulation estimator for estimating an amplitude modulation or a frequency modulation or a phase modulation for each band of the plurality of band pass filters for the portion of the audio signal using the information on the plurality of band pass filters; and an output interface for transmitting, storing or modifying information on the amplitude modulation, information on the frequency modulation or phase modulation or the information on the plurality of band pass filters for the portion of the audio signal.

摘要翻译： 用于将音频信号转换为参数化表示的装置具有用于分析音频信号的一部分以获得分析结果的信号分析器; 用于基于分析结果估计多个带通滤波器的信息的带通估计器，其中关于所述多个带通滤波器的信息具有关于所述音频信号的所述部分的滤波器形状的信息，其中， 带通滤波器在音频频谱上是不同的，取决于带通滤波器的中心频率; 调制估计器，用于使用关于所述多个带通滤波器的信息来估计用于所述音频信号的所述部分的所述多个带通滤波器的每个频带的幅度调制或频率调制或相位调制; 以及用于发送，存储或修改关于振幅调制的信息，关于频率调制或相位调制的信息或关于音频信号的该部分的多个带通滤波器的信息的输出接口。

公开(公告)号：US20100198586A1

公开(公告)日：2010-08-05

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： G10L19/02

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。

公开(公告)号：US07720230B2

公开(公告)日：2010-05-18

申请号：US11006482

申请日：2004-12-07

申请人： Eric Allamanche ,  Sascha Disch ,  Christof Faller ,  Juergen Herre

发明人： Eric Allamanche ,  Sascha Disch ,  Christof Faller ,  Juergen Herre

IPC分类号： H04R5/00

CPC分类号： G10L19/008

摘要： At an audio encoder, cue codes are generated for one or more audio channels, wherein an envelope cue code is generated by characterizing a temporal envelope in an audio channel. At an audio decoder, E transmitted audio channel(s) are decoded to generate C playback audio channels, where C>E≧1. Received cue codes include an envelope cue code corresponding to a characterized temporal envelope of an audio channel corresponding to the transmitted channel(s). One or more transmitted channel(s) are upmixed to generate one or more upmixed channels. One or more playback channels are synthesized by applying the cue codes to the one or more upmixed channels, wherein the envelope cue code is applied to an upmixed channel or a synthesized signal to adjust a temporal envelope of the synthesized signal based on the characterized temporal envelope such that the adjusted temporal envelope substantially matches the characterized temporal envelope.

摘要翻译： 在音频编码器中，为一个或多个音频通道生成提示码，其中通过表征音频通道中的时间包络来产生包络线索码。 在音频解码器处，对E个发送的音频信道进行解码以生成C个回放音频信道，其中C>E≥1。 接收的提示码包括与对应于所发送的频道的音频信道的特征化时间包络对应的信封提示码。 一个或多个传输的信道被混合以产生一个或多个上混频道。 通过将提示码应用于一个或多个上混合通道来合成一个或多个回放通道，其中，将包络提示码应用于上混合通道或合成信号，以基于表征的时间包络线来调整合成信号的时间包络 使得调整的时间包络基本上与所表征的时间包络相匹配。

公开(公告)号：US20080027729A1

公开(公告)日：2008-01-31

申请号：US11554492

申请日：2006-10-30

申请人： Juergen Herre ,  Ralph Kulessa ,  Sascha Disch ,  Karsten Linzmeier ,  Christian Neubauer ,  Frank Siebenhaar

发明人： Juergen Herre ,  Ralph Kulessa ,  Sascha Disch ,  Karsten Linzmeier ,  Christian Neubauer ,  Frank Siebenhaar

IPC分类号： G10L21/00

CPC分类号： H04H20/31

摘要： According to an inventive scheme for introducing a watermark into an information signal, the information signal is at first transferred from a time representation to a spectral/modulation spectral representation). The information signal is then manipulated in the spectral/modulation spectral representation in dependence on the watermark to be introduced to obtain a modified spectral/modulation spectral representation, and subsequently an information signal provided with a watermark is formed based on the modified spectral/modulation spectral representation. An advantage is that, due to the fact that the watermark is embedded and/or derived in the spectral/modulation spectral representation or range, traditional correlation attacks as are used in watermark methods based on a spread-band modulation cannot succeed easily.

摘要翻译： 根据用于将水印引入信息信号的发明方案，信息信号首先从时间表示传送到频谱/调制频谱表示）。 然后根据要引入的水印在频谱/调制频谱表示中操作信息信号以获得修改的频谱/调制频谱表示，随后基于修改的频谱/调制频谱形成提供有水印的信息信号 表示。 优点在于，由于在频谱/调制频谱表示或范围内嵌入和/或导出水印的事实，所以在基于扩频调制的水印方法中使用的传统的相关攻击不能容易地成功。

公开(公告)号：US20070276889A1

公开(公告)日：2007-11-29

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：US20070236858A1

公开(公告)日：2007-10-11

申请号：US11384000

申请日：2006-05-18

申请人： Sascha Disch ,  Karsten Linzmeier ,  Juergen Herre ,  Harald Popp

发明人： Sascha Disch ,  Karsten Linzmeier ,  Juergen Herre ,  Harald Popp

IPC分类号： H01G4/255

CPC分类号： H04S3/002 ,  G10L19/008 ,  G10L19/26 ,  H04R2217/03 ,  H04S3/02 ,  H04S2420/03

摘要： The present invention is based on the finding that a reconstructed output channel, reconstructed with a multi-channel reconstructor using at least one downmix channel derived by downmixing a plurality of original channels and using a parameter representation including additional information on a temporal fine structure of an original channel can be reconstructed efficiently with high quality, when a generator for generating a direct signal component and a diffuse signal component based on the downmix channel is used. The quality can be essentially enhanced, if only the direct signal component is modified such that the temporal fine structure of the reconstructed output channel is fitting a desired temporal fine structure, indicated by the additional information on the temporal fine structure transmitted.

摘要翻译： 本发明基于以下发现：重建的输出通道，其使用通过将多个原始通道进行下混合而导出的至少一个下混频道重建的多通道重构器，并且使用参数表示，该参数表示包括关于时间精细结构的附加信息 当使用用于产生基于下混通道的直接信号分量和扩散信号分量的发生器时，可以高质量地重构原始信道。 如果仅修改直接信号分量，使得重建的输出信道的时间精细结构适合于由发送的时间精细结构的附加信息所指示的期望的时间精细结构，则质量可以基本上增强。

公开(公告)号：US20070081597A1

公开(公告)日：2007-04-12

申请号：US11363985

申请日：2006-02-27

申请人： Sascha Disch ,  Juergen Herre ,  Matthias Neusinger ,  Dirk Jeroen Breebaart ,  Gerard Hotho

发明人： Sascha Disch ,  Juergen Herre ,  Matthias Neusinger ,  Dirk Jeroen Breebaart ,  Gerard Hotho

IPC分类号： H04B14/04

CPC分类号： G10L19/008 ,  H04S3/008

摘要： A selected channel of a multi-channel signal which is represented by frames composed from sampling values having a high time resolution can be encoded with higher quality when a wave form parameter representation representing a wave form of an intermediate resolution representation of the selected channel is derived, the wave form parameter representation including a sequence of intermediate wave form parameters having a time resolution lower than the high time resolution of the sampling values and higher than a time resolution defined by a frame repetition rate. The wave form parameter representation with the intermediate resolution can be used to shape a reconstructed channel to retrieve a channel having a signal envelope close to that one of the selected original channel. The time scale on which the shaping is performed is shorter than the time scale of a framewise processing, thus enhancing the quality of the reconstructed channel. On the other hand, the shaping time scale is larger than the time scale of the sampling values, significantly reducing the amount of data needed by the wave form parameter representation.

摘要翻译： 当表示所选频道的中间分辨率表示的波形的波形参数表示被导出时，由具有高时间分辨率的采样值组成的帧表示的多信道信号的选择信道可以被更高质量地编码 波形参数表示，包括具有低于采样值的高时间分辨率的时间分辨率并且高于由帧重复率定义的时间分辨率的中间波形参数序列。 具有中间分辨率的波形参数表示可用于对重建的信道进行整形以检索具有接近所选择的原始信道中的那一个的信号包络的信道。 进行整形的时间标度比框架处理的时间标度短，从而提高重构信道的质量。 另一方面，成形时间尺度大于采样值的时间尺度，显着减少波形参数表示所需的数据量。

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。

公开(公告)号：US08855320B2

公开(公告)日：2014-10-07

申请号：US13291986

申请日：2011-11-08

申请人： Sascha Disch ,  Ville Pulkki ,  Mikko-Ville Laitinen ,  Cumhur Erkut

发明人： Sascha Disch ,  Ville Pulkki ,  Mikko-Ville Laitinen ,  Cumhur Erkut

IPC分类号： H04R5/00 ,  H04S7/00

CPC分类号： H04S7/30 ,  H04S2400/11 ,  H04S2420/03

摘要： An apparatus for determining a spatial output multi-channel audio signal based on an input audio signal and an input parameter. The apparatus includes a decomposer for decomposing the input audio signal based on the input parameter to obtain a first decomposed signal and a second decomposed signal different from each other. Furthermore, the apparatus includes a renderer for rendering the first decomposed signal to obtain a first rendered signal having a first semantic property and for rendering the second decomposed signal to obtain a second rendered signal having a second semantic property being different from the first semantic property. The apparatus comprises a processor for processing the first rendered signal and the second rendered signal to obtain the spatial output multi-channel audio signal.

摘要翻译： 一种用于基于输入音频信号和输入参数来确定空间输出多声道音频信号的装置。 该装置包括：分解器，用于基于输入参数分解输入音频信号，以获得彼此不同的第一分解信号和第二分解信号。 此外，该装置包括用于渲染第一分解信号以获得具有第一语义属性的第一渲染信号的渲染器，并且用于渲染第二分解信号以获得具有与第一语义属性不同的第二语义属性的第二渲染信号。 该装置包括用于处理第一渲染信号和第二渲染信号的处理器，以获得空间输出多声道音频信号。