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1. 发明申请

WO2012110473A1 APPARATUS AND METHOD FOR ENCODING AND DECODING AN AUDIO SIGNAL USING AN ALIGNED LOOK-AHEAD PORTION 审中-公开
标题翻译：使用对齐的前置部分编码和解码音频信号的装置和方法
公开(公告)号：WO2012110473A1
公开(公告)日：2012-08-23
申请号：PCT/EP2012/052450
申请日：2012-02-14
申请人： FRAUNHOFER-GESELLSCHAFT ZUR FÖRDERUNG DER ANGEWANDTEN FORSCHUNG E.V. , RAVELLI, Emmanuel , GEIGER, Ralf , SCHNELL, Markus , FUCHS, Guillaume , RUOPPILA, Vesa , BÄCKSTRÖM, Tom , GRILL, Bernhard , HELMRICH, Christian
发明人： RAVELLI, Emmanuel , GEIGER, Ralf , SCHNELL, Markus , FUCHS, Guillaume , RUOPPILA, Vesa , BÄCKSTRÖM, Tom , GRILL, Bernhard , HELMRICH, Christian
IPC分类号： G10L19/02 , G10L19/04 , G10L19/00 , G10L19/14
CPC分类号： G10L19/00 , G10K11/16 , G10L19/005 , G10L19/012 , G10L19/0212 , G10L19/025 , G10L19/03 , G10L19/04 , G10L19/107 , G10L19/12 , G10L19/22 , G10L19/26 , G10L21/0216 , G10L25/06 , G10L25/78
摘要： An apparatus for encoding an audio signal having a stream of audio samples (100) comprises: a windower (102) for applying a prediction coding analysis window (200) to the stream of audio samples to obtain windowed data for a prediction analysis and for applying a transform coding analysis window (204) to the stream of audio samples to obtain windowed data for a transform analysis, wherein the transform coding analysis window is associated with audio samples within a current frame of audio samples and with audio samples of a predefined portion of a future frame of audio samples being a transform-coding look-ahead portion (206), wherein the prediction coding analysis window is associated with at least the portion of the audio samples of the current frame and with audio samples of a predefined portion of the future frame being a prediction coding look-ahead portion (208), wherein the transform coding look-ahead portion (206) and the prediction coding look-ahead portion (208) are identically to each other or are different from each other by less than 20% of the prediction coding look-ahead portion (208) or less than 20% of the transform coding look-ahead portion 206; and an encoding processor (104) for generating prediction coded data for the current frame using the windowed data for the prediction analysis or for generating transform coded data for the current frame using the windowed data for the transform analysis.
摘要翻译：一种用于对具有音频样本流（100）的音频信号进行编码的装置包括：用于将预测编码分析窗口（200）应用于音频样本流的收敛机（102），以获得用于预测分析和应用的窗口数据变换编码分析窗口（204）到音频样本流以获得用于变换分析的加窗数据，其中变换编码分析窗口与音频样本的当前帧内的音频样本相关联，并且具有预定义部分的音频样本音频样本的未来帧是变换编码先行部分（206），其中预测编码分析窗口与当前帧的音频样本的至少一部分相关联，并且与预定义部分的音频样本相关联未来帧是预测编码预测部分（208），其中变换编码预测部分（206）和预测编码预览部分（208）是相同的彼此相差不到预测编码预测部分（208）的20％或小于变换编码预读部分206的20％; 以及编码处理器（104），用于使用用于预测分析的加窗数据产生用于当前帧的预测编码数据，或者使用用于变换分析的窗口数据来生成当前帧的变换编码数据。

2. 发明申请

WO2011147950A1 LOW-DELAY UNIFIED SPEECH AND AUDIO CODEC 审中-公开
标题翻译：低延迟语音和音频编解码器
公开(公告)号：WO2011147950A1
公开(公告)日：2011-12-01
申请号：PCT/EP2011/058701
申请日：2011-05-27
申请人： FRAUNHOFER-GESELLSCHAFT ZUR FÖRDERUNG DER ANGEWANDTEN FORSCHUNG E.V. , GEIGER, Ralf , SCHNELL, Markus , FUCHS, Guillaume , RAVELLI, Emmanuel , BÄCKSTRÖM, Tom , LECOMTE, Jérémie , SCHMIDT, Konstantin , RETTELBACH, Nikolaus , LUTZKY, Manfred , GRILL, Bernhard
发明人： GEIGER, Ralf , SCHNELL, Markus , FUCHS, Guillaume , RAVELLI, Emmanuel , BÄCKSTRÖM, Tom , LECOMTE, Jérémie , SCHMIDT, Konstantin , RETTELBACH, Nikolaus , LUTZKY, Manfred , GRILL, Bernhard
IPC分类号： G10L19/14
CPC分类号： G10L19/20
摘要： A unified speech and audio decoder is described, which comprises a frame buffer configured to buffer a sub-part of a datastream composed of consecutive frames in units of the frames so that the subpart continuously comprises at least one frame, each frame representing a coded version of a respective portion of consecutive portions of an audio signal, and each frame comprising a mode identifier assigning the respective frame to a respective one of a plurality of coding modes comprising a CELP (codebook excitation linear prediction) coding mode and a transform coded excitation linear prediction coding mode. Further, the unified speech and audio decoder comprises a CELP decoder configured to decode the frames to which the CELP coding mode is assigned to reconstruct the respective portions, and a transform coded excitation linear prediction decoder configured to decode the frames to which the transform coded excitation linear prediction coding mode is assigned, to reconstruct the respective portions, wherein the frame buffer is configured to distribute the frames buffered to the CELP decoder and the transform coded excitation linear prediction decoder under removal of the respective frames from the frame buffer, frame-wise.
摘要翻译：描述了一种统一的语音和音频解码器，其包括帧缓冲器，其被配置为以帧为单位缓冲由连续帧组成的数据流的子部分，使得该子部分连续地包括至少一个帧，每个帧表示编码版本音频信号的连续部分的相应部分，并且每个帧包括将各个帧分配给包括CELP（码本激励线性预测）编码模式和变换编码激励线性的多个编码模式中的相应一个的模式标识符预测编码模式。此外，统一语音和音频解码器包括：CELP解码器，被配置为对分配有CELP编码模式的帧进行解码以重构各个部分;以及变换编码激励线性预测解码器，被配置为对转换编码激励分配线性预测编码模式，以重建各个部分，其中帧缓冲器被配置为在从帧缓冲器移除相应帧的情况下，将缓冲到CELP解码器和变换编码激励线性预测解码器的帧分配。

3. 发明申请

WO2011110594A1 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 审中-公开
标题翻译：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：WO2011110594A1
公开(公告)日：2011-09-15
申请号：PCT/EP2011/053541
申请日：2011-03-09
申请人： FRAUNHOFER-GESELLSCHAFT ZUR FÖRDERUNG DER ANGEWANDTEN FORSCHUNG E.V. , DOLBY INTERNATIONAL AB , BAYER, Stefan , BÄCKSTRÖM, Tom , GEIGER, Ralf , EDLER, Bernd , DISCH, Sascha , VILLEMOES, Lars
发明人： BAYER, Stefan , BÄCKSTRÖM, Tom , GEIGER, Ralf , EDLER, Bernd , DISCH, Sascha , VILLEMOES, Lars
IPC分类号： G10L21/04
CPC分类号： G10L19/022 , G10L19/0212 , G10L25/90
摘要： An audio signal decoder (150 for providing a decoded audio signal representation (154) on the basis of an encoded audio signal representation (152) comprising an encoded spectrum representation (ac_spectral_data[]) and an encoded time warp information (tw_data[]) comprises a context-based spectral value decoder (160) configured to decode a codeword (acod_m) describing one or more spectral values or at least a portion (m) of a number representation of one or more spectral values in dependence on a context state, to obtain decoded spectral values (162, 297, x_ac_dec[]). The audio signal decoder also comprises a context state determinator (170) configured to determine a current context state (164, c) in dependence on one or more previously decoded spectral values (162, 297). The audio signal decoder also comprises a time warping frequency-domain-to-time-domain converter (180) configured to provide a time-warped time-domain representation (182) of a given audio frame on the basis of a set of decoded spectral values (162) associated with the given audio frame and provided by the context-based spectral value decoder and in dependence on the time warp information. The context-state determinator (170) 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（150）用于基于包括编码频谱表示（ac_spectral_data []）和编码时间扭曲信息（tw_data []）的编码音频信号表示（152）提供解码音频信号表示（154）基于上下文的频谱值解码器（160），被配置为对描述根据上下文状态的一个或多个频谱值的数字表示的一个或多个频谱值或至少一部分（m）的码字（acod_m）进行解码，音频信号解码器还包括上下文状态确定器（170），配置为根据一个或多个先前解码的频谱值（164，c）来确定当前上下文状态（164，c）音频信号解码器还包括时间扭曲频域 - 时域转换器（180），其被配置为基于以下方式提供给定音频帧的时间扭曲时域表示（182）一套与由给定音频帧相关联并由基于上下文的频谱值解码器提供并且依赖于时间扭曲信息的解码频谱值（162）。上下文状态确定器（170）被配置为使上下文状态的确定适应于后续音频帧之间的基本频率的改变。音频信号编码器应用可比较的概念。

4. 发明申请

WO2010084046A1 APPARATUS, METHOD AND COMPUTER PROGRAM FOR OBTAINING A PARAMETER DESCRIBING A VARIATION OF A SIGNAL CHARACTERISTIC OF A SIGNAL 审中-公开
标题翻译：用于获取描述信号信号特征变化的参数的装置，方法和计算机程序
公开(公告)号：WO2010084046A1
公开(公告)日：2010-07-29
申请号：PCT/EP2010/050229
申请日：2010-01-11
申请人： FRAUNHOFER-GESELLSCHAFT ZUR FÖRDERUNG DER ANGEWANDTEN FORSCHUNG E.V. , BÄCKSTRÖM, Tom , BAYER, Stefan , GEIGER, Ralf , NEUENDORF, Max , DISCH, Sascha
发明人： BÄCKSTRÖM, Tom , BAYER, Stefan , GEIGER, Ralf , NEUENDORF, Max , DISCH, Sascha
IPC分类号： G10L11/04
CPC分类号： G10L25/90
摘要： An apparatus for obtaining a parameter describing a variation of a signal characteristic of a signal on the basis of actual transform-domain parameters describing the audio signal in transform-domain comprises a parameter determinator. The parameter determinator is configured to determine one or more model parameters of a transform- domain variation model describing an evolution of the transform-domain parameters in dependence on one or more model parameters representing a signal characteristic.
摘要翻译：根据描述变换域中的音频信号的实际变换域参数，获得描述信号的信号特性变化的参数的装置包括参数确定器。参数确定器被配置为根据表示信号特性的一个或多个模型参数来确定描述变换域参数的演变的变换域变体模型的一个或多个模型参数。

5. 发明申请

WO2011110591A1 AUDIO SIGNAL DECODER, AUDIO SIGNAL ENCODER, METHODS AND COMPUTER PROGRAM USING A SAMPLING RATE DEPENDENT TIME-WARP CONTOUR ENCODING 审中-公开
标题翻译：音频信号解码器，音频信号编码器，使用采样速率的方法和计算机程序依赖于时间波形编码
公开(公告)号：WO2011110591A1
公开(公告)日：2011-09-15
申请号：PCT/EP2011/053538
申请日：2011-03-09
申请人： FRAUNHOFER-GESELLSCHAFT ZUR FÖRDERUNG DER ANGEWANDTEN FORSCHUNG E.V. , DOLBY INTERNATIONAL AB , BAYER, Stefan , BÄCKSTRÖM, Tom , GEIGER, Ralf , EDLER, Bernd , DISCH, Sascha , VILLEMOES, Lars
发明人： BAYER, Stefan , BÄCKSTRÖM, Tom , GEIGER, Ralf , EDLER, Bernd , DISCH, Sascha , VILLEMOES, Lars
IPC分类号： G10L19/02
CPC分类号： G10L19/022 , G10L19/0212 , G10L25/90
摘要： An audio signal decoder configured to provide a decoded audio signal representation on the basis of an encoded audio signal representation comprising a sampling frequency information, an encoded time warp information and an encoded spectrum representation comprises a time warp calculator and a warp decoder. The time warp calculator is configured to adapt a mapping rule for mapping codewords of the encoded time warp information onto decoded time warp values describing the decoded time warp information in dependence on the sampling frequency information. The warp decoder is configured to provide the decoded audio signal representation on the basis of the encoded spectrum representation and in dependence on the decoded time warp information.
摘要翻译：音频信号解码器被配置为基于包括采样频率信息，编码时间扭曲信息和编码频谱表示的编码音频信号表示来提供解码音频信号表示，包括时间扭曲计算器和扭曲解码器。时间扭曲计算器被配置为根据编码时间扭曲信息的码字将映射规则适应于根据采样频率信息描述解码的时间扭曲信息的解码时间扭曲值。翘曲解码器被配置为基于编码的频谱表示并且依赖于解码的时间扭曲信息来提供解码的音频信号表示。

6. 发明申请

WO2013024085A1 OPTIMAL MIXING MATRICES AND USAGE OF DECORRELATORS IN SPATIAL AUDIO PROCESSING 审中-公开
标题翻译：空间混合矩阵和装饰在空间音频处理中的应用
公开(公告)号：WO2013024085A1
公开(公告)日：2013-02-21
申请号：PCT/EP2012/065861
申请日：2012-08-14
申请人： FRAUNHOFER-GESELLSCHAFT ZUR FÖRDERUNG DER ANGEWANDTEN FORSCHUNG E.V. , VILKAMO, Juha , BÄCKSTRÖM, Tom , KÜCH, Fabian , KUNTZ, Achim
发明人： VILKAMO, Juha , BÄCKSTRÖM, Tom , KÜCH, Fabian , KUNTZ, Achim
IPC分类号： G10L19/00
CPC分类号： G10H1/183 , G10L19/008
摘要： An apparatus for generating an audio output signal having two or more audio output channels from an audio input signal having two or more audio input channels is provided. The apparatus comprises a provider (110) and a signal processor (120). The provider (110) is adapted to provide first covariance properties of the audio input signal. The signal processor (120) is adapted to generate the audio output signal by applying a mixing rule on at least two of the two or more audio input channels. The signal processor (120) is configured to determine the mixing rule based on the first covariance properties of the audio input signal and based on second covariance properties of the audio output signal, the second covariance properties being different from the first covariance properties.
摘要翻译：提供一种用于从具有两个或多个音频输入通道的音频输入信号产生具有两个或多个音频输出通道的音频输出信号的装置。该装置包括提供商（110）和信号处理器（120）。提供者（110）适于提供音频输入信号的第一协方差属性。信号处理器（120）适于通过在两个或多个音频输入通道中的至少两个上施加混合规则来产生音频输出信号。信号处理器（120）被配置为基于音频输入信号的第一协方差特性并基于音频输出信号的第二协方差属性来确定混合规则，第二协方差属性不同于第一协方差属性。

7. 发明申请

WO2017050972A1 ENCODER AND METHOD FOR ENCODING AN AUDIO SIGNAL WITH REDUCED BACKGROUND NOISE USING LINEAR PREDICTIVE CODING 审中-公开
标题翻译：使用线性预测编码的减少背景噪声编码音频信号的编码器和方法
公开(公告)号：WO2017050972A1
公开(公告)日：2017-03-30
申请号：PCT/EP2016/072701
申请日：2016-09-23
申请人： FRAUNHOFER-GESELLSCHAFT ZUR FÖRDERUNG DER ANGEWANDTEN FORSCHUNG E.V. , FRIEDRICH-ALEXANDER UNIVERSITAET ERLANGEN-NUERNBERG
发明人： FISCHER, Johannes , BÄCKSTRÖM, Tom , JOKINEN, Emma
IPC分类号： G10L19/06 , G10L21/0208
CPC分类号： G10L19/06 , G10L21/0208
摘要： It is shown an encoder for encoding an audio signal with reduced background noise using linear predictive coding. The encoder comprises a background noise estimator configured to estimate background noise of the audio signal, a background noise reducer configured to generate background noise reduced audio signal by subtracting the estimated background noise of the audio signal from the audio signal, and a predictor configured to subject the audio signal to linear prediction analysis to obtain a first set of linear prediction filter (LPC) coefficients and to subject the background noise reduced audio signal to linear prediction analysis to obtain a second set of linear prediction filter (LPC) coefficients. Furthermore, the encoder comprises an analysis filter composed of a cascade of time-domain filters controlled by the obtained first set of LPC coefficients and the obtained second set of LPC coefficients.
摘要翻译：示出了使用线性预测编码对具有降低的背景噪声的音频信号进行编码的编码器。编码器包括被配置为估计音频信号的背景噪声的背景噪声估计器，被配置为通过从音频信号中减去音频信号的估计背景噪声来生成背景噪声降低音频信号的背景噪声减少器，以及被配置为对象音频信号进行线性预测分析，以获得第一组线性预测滤波器（LPC）系数，并将背景噪声降低的音频信号进行线性预测分析，以获得第二组线性预测滤波器（LPC）系数。此外，编码器包括由所获得的第一组LPC系数控制的级联的时域滤波器和所获得的第二组LPC系数组成的分析滤波器。

8. 发明申请

WO2015132048A1 CONCEPT FOR ENCODING OF INFORMATION 审中-公开
标题翻译：概念编码信息
公开(公告)号：WO2015132048A1
公开(公告)日：2015-09-11
申请号：PCT/EP2015/052634
申请日：2015-02-09
申请人： FRAUNHOFER-GESELLSCHAFT ZUR FÖRDERUNG DER ANGEWANDTEN FORSCHUNG E.V. , FRIEDRICH-ALEXANDER-UNIVERSITAET ERLANGEN-NUERNBERG
发明人： BÄCKSTRÖM, Tom , FISCHER PEDERSEN, Christian , FISCHER, Johannes , HÜTTENBERGER, Matthias , PINO, Alfonso
IPC分类号： G10L19/07
CPC分类号： G10L19/07 , G10L19/0212 , G10L19/038 , G10L2019/0011 , G10L2019/0016
摘要： The invention provides an information encoder for encoding an information signal (IS), the information encoder (1) comprising: an analyzer (2) for analyzing the information signal (IS) in order to obtain linear prediction coefficients of a predictive polynomial A(z); a converter (3) for converting the linear prediction coefficients of the predictive polynomial A(z) to frequency values f 1 ...f n of a spectral frequency representation of the predictive polynomial A(z), wherein the converter (3) is configured to determine the frequency values f 1 ...f n by analyzing a pair of polynomials P(z) and Q(z) being defined as P(z) = A(z) + z -m-l A(z -1 ) and Q(z) = A(z) - z -m-l A(z -1 ), wherein m is an order of the predictive polynomial A(z) and I is greater or equal to zero, wherein the converter (3) is configured to obtain the frequency values (f 1 ...f n ) by establishing a strictly real spectrum (RES) derived from P(z) and a strictly imaginary spectrum (IES) from Q(z) and by identifying zeros of the strictly real spectrum (RES) derived from P(z) and the strictly imaginary spectrum (IES) derived from Q(z); a quantizer (4) for obtaining quantized frequency (f q1 ...f qn ) values from the frequency values (f 1 ...f n ); and a bitstream producer (5) for producing a bitstream comprising the quantized frequency values (f q1 ...f qn ).
摘要翻译：本发明提供了一种用于对信息信号（IS）进行编码的信息编码器，该信息编码器（1）包括：分析器（2），用于分析信息信号（IS），以获得预测多项式A（z）的线性预测系数）; 用于将预测多项式A（z）的线性预测系数转换成预测多项式A（z）的频谱频率表示的频率值f1 ... fn的转换器（3），其中转换器（3）被配置为通过分析一对多项式P（z）和Q（z）被定义为P（z）= A（z）+ zm-1A（z-1）和Q（z）来确定频率值f1 ... fn， = A（z）-zm-1A（z-1），其中m是预测多项式A（z）的阶数，I大于或等于零，其中转换器（3）被配置为获得频率值通过建立从P（z）导出的严格实数谱（RES）和来自Q（z）的严格虚数谱（IES），并通过识别从P（z）导出的严格实数谱（RES）（z）和衍生自Q（z）的严格虚数谱（IES）; 用于从频率值（f1 ... fn）获得量化频率（fq1 ... fqn）的量化器（4）; 以及用于产生包括量化频率值（fq1 ... fqn）的比特流的比特流产生器（5）。

9. 发明申请

WO2015135797A1 ENCODER, DECODER AND METHOD FOR ENCODING AND DECODING 审中-公开
标题翻译：编码器，解码器和编码和解码的方法
公开(公告)号：WO2015135797A1
公开(公告)日：2015-09-17
申请号：PCT/EP2015/054396
申请日：2015-03-03
申请人： FRAUNHOFER-GESELLSCHAFT ZUR FÖRDERUNG DER ANGEWANDTEN FORSCHUNG E.V. , FRIEDRICH-ALEXANDER-UNIVERSITAET ERLANGEN-NUERNBERG
发明人： BÄCKSTRÖM, Tom , FISCHER, Johannes Karl , HELMRICH, Christian
IPC分类号： G10L19/08
CPC分类号： G10L19/167 , G10L19/0212 , G10L19/028 , G10L19/032 , G10L19/08
摘要： An encoder for encoding an audio signal into a data stream comprises a predictor, a factorizer, a transformer and a quantize and encode stage. The predictor is configured to analyze the audio signal in order to obtain prediction coefficients describing a spectral analog of the audio signal or a fundamental frequency of the audio signal and subject the audio signal to an analysis filter function dependent on the prediction coefficients in order to output a residual signal of the audio signal. The factorizer is configured to apply a matrix factorization onto an audiocorrelation or covariance matrix of synthesis filter function defined by the prediction coefficients to obtain factorized matrices. The transformer is configured to transform the residual signal based on the factorized matrices to obtain a transformed residual signal. The quantize and decode stage is configured to quantize the transformed residual signal to obtain a quantized transformed residual signal or an encoded quantized transformed residual signal.
摘要翻译：用于将音频信号编码成数据流的编码器包括预测器，分解器，变压器和量化和编码级。预测器被配置为分析音频信号，以便获得描述音频信号的频谱模拟或音频信号的基频的预测系数，并将该音频信号根据预测系数进行分析滤波器功能，以输出音频信号的残留信号。因子分解器被配置为将矩阵分解应用于由预测系数定义的合成滤波函数的音频相关或协方差矩阵，以获得因子化矩阵。变压器被配置为基于因式分解矩阵来变换残差信号以获得经变换的残差信号。量化和解码级被配置为量化变换的残差信号以获得经量化的变换残差信号或编码的量化变换残差信号。

10. 发明申请

WO2017157800A1 ENCODING APPARATUS FOR PROCESSING AN INPUT SIGNAL AND DECODING APPARATUS FOR PROCESSING AN ENCODED SIGNAL 审中-公开
标题翻译：用于处理输入信号的编码设备和用于处理编码信号的解码设备
公开(公告)号：WO2017157800A1
公开(公告)日：2017-09-21
申请号：PCT/EP2017/055716
申请日：2017-03-10
申请人： FRAUNHOFER-GESELLSCHAFT ZUR FÖRDERUNG DER ANGEWANDTEN FORSCHUNG E.V.
发明人： BÄCKSTRÖM, Tom , GHIDO, Florin , FISCHER, Johannes
IPC分类号： G10L19/038
摘要： The invention refers to an apparatus (1) for processing an input signal (3), comprising a perceptual weighter (10) and a quantizer (14). The perceptual weighter (10) comprises a model provider (12) and a model applicator (13). The model provider (12) provides a perceptual weighted model (W) based on the input signal (3). The model applicator (13) provides a perceptually weighted spectrum (x) by applying the perceptual weighted model (W) to a spectrum (s) based on the input signal (3). The quantizer (14) is configured to quantize the perceptually weighted spectrum (x) and for providing a bitstream (fi). The quantizer (14) comprises a random matrix applicator (16) and a sign function calculator (17). The random matrix applicator (16) is configured for applying a random matrix (P) to the perceptually weighted spectrum (x) in order to provide a transformed spectrum (u). The sign function calculator (17) is configured for calculating a sign function of components of the transformed spectrum (u) in order to provide the bitstream (u). The invention further refers to an apparatus (2) for processing an encoded signal (4) and to corresponding methods.
摘要翻译：本发明涉及一种用于处理输入信号（3）的设备（1），其包括感知加权器（10）和量化器（14）。知觉调节器（10）包括模型提供器（12）和模型施加器（13）。模型提供者（12）基于输入信号（3）提供感知加权模型（W）。模型施加器（13）通过基于输入信号（3）将感知加权模型（W）应用于谱来提供感知加权谱（x）。量化器（14）被配置为量化感知加权谱（x）并提供比特流（fi）。量化器（14）包括随机矩阵施加器（16）和符号函数计算器（17）。随机矩阵施加器（16）被配置用于将随机矩阵（P）应用于感知加权谱（x）以提供经变换的谱（u）。符号函数计算器（17）被配置用于计算变换频谱（u）的分量的符号函数以便提供比特流（u）。本发明还涉及一种用于处理编码信号（4）的装置（2）和相应的方法。

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