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11. 发明申请

US20200013417A1 COMFORT NOISE ADDITION FOR MODELING BACKGROUND NOISE AT LOW BIT-RATES 审中-公开
公开(公告)号：US20200013417A1
公开(公告)日：2020-01-09
申请号：US16448291
申请日：2019-06-21
申请人： Fraunhofer-Gesellschaft zur Foerderung der angewandten Forschung e.V.
发明人： Guillaume FUCHS , Anthony LOMBARD , Emmanuel RAVELLI , Stefan DOEHLA , Jérémie LECOMTE , Martin DIETZ
IPC分类号： G10L19/012
摘要： The invention provides a decoder being configured for processing an encoded audio bitstream, wherein the decoder includes: a bitstream decoder configured to derive a decoded audio signal from the bitstream, wherein the decoded audio signal includes at least one decoded frame; a noise estimation device configured to produce a noise estimation signal containing an estimation of the level and/or the spectral shape of a noise in the decoded audio signal; a comfort noise generating device configured to derive a comfort noise signal from the noise estimation signal; and a combiner configured to combine the decoded frame of the decoded audio signal and the comfort noise signal in order to obtain an audio output signal.

12. 发明申请

US20180322884A1 Apparatus and Method for Estimating an Inter-Channel Time Difference 审中-公开
公开(公告)号：US20180322884A1
公开(公告)日：2018-11-08
申请号：US16035456
申请日：2018-07-13
申请人： Fraunhofer-Gesellschaft zur Foerderung der angewandten Forschung e.V.
发明人： Stefan BAYER , Eleni FOTOPOULOU , Markus MULTRUS , Guillaume FUCHS , Emmanuel RAVELLI , Markus SCHNELL , Stefan DOEHLA , Wolfgang JAEGERS , Martin DIETZ , Goran MARKOVIC
IPC分类号： G10L19/008 , G10L19/022 , G10L19/04
摘要： An apparatus for estimating an inter-channel time difference between a first channel signal and a second channel signal, includes: a calculator for calculating a cross-correlation spectrum for a time block from the first channel signal in the time block and the second channel signal in the time block; a spectral characteristic estimator for estimating a characteristic of a spectrum of the first channel signal or the second channel signal for the time block; a smoothing filter for smoothing the cross-correlation spectrum over time using the spectral characteristic to obtain a smoothed cross-correlation spectrum; and a processor for processing the smoothed cross-correlation spectrum to obtain the inter-channel time difference.

13. 发明申请

US20180197552A1 Apparatus and Method for Encoding or Decoding a Multi-Channel Signal Using Spectral-Domain Resampling 审中-公开
公开(公告)号：US20180197552A1
公开(公告)日：2018-07-12
申请号：US15821108
申请日：2017-11-22
申请人： Fraunhofer-Gesellschaft zur Foerderung der angewandten Forschung e.V.
发明人： Guillaume FUCHS , Emmanuel RAVELLI , Markus MULTRUS , Markus SCHNELL , Stefan DOEHLA , Martin DIETZ , Goran MARKOVIC , Eleni FOTOPOULOU , Stefan BAYER , Wolfgang JAEGERS
IPC分类号： G10L19/008 , G10L19/022 , G10L19/04
摘要： An apparatus for encoding a multi-channel signal having at least two channels, has: a time-spectral converter for converting sequences of blocks of sample values of the at least two channels into a frequency domain representation having sequences of blocks of spectral values for the at least two channels, wherein a block of sampling values has an associated input sampling rate, and a block of spectral values of the sequences of blocks of spectral values has spectral values up to a maximum input frequency being related to the input sampling rate; a multi-channel processor to obtain at least one result sequence of blocks of spectral values having information related to the at least two channels; a spectral domain resampler to obtain a resampled sequence of blocks of spectral values; a spectral-time converter for converting the resampled sequence of blocks of spectral values into a time domain representation; and a core encoder for encoding the output sequence of blocks of sampling values to obtain an encoded multi-channel signal.

14. 发明申请

US20170365264A1 AUDIO ENCODER FOR ENCODING A MULTICHANNEL SIGNAL AND AUDIO DECODER FOR DECODING AN ENCODED AUDIO SIGNAL 审中-公开
公开(公告)号：US20170365264A1
公开(公告)日：2017-12-21
申请号：US15695668
申请日：2017-09-05
申请人： Fraunhofer-Gesellschaft zur Foerderung der angewandten Forschung e.V.
发明人： Sascha DISCH , Guillaume FUCHS , Emmanuel RAVELLI , Christian NEUKAM , Konstantin SCHMIDT , Conrad BENNDORF , Andreas NIEDERMEIER , Benjamin SCHUBERT , Ralf GEIGER
IPC分类号： G10L19/008 , G10L19/032 , G10L19/13 , G10L19/18
CPC分类号： G10L19/008 , G10L19/02 , G10L19/032 , G10L19/04 , G10L19/13 , G10L19/18 , G10L21/038
摘要： Audio encoder for encoding a multichannel signal is shown. The audio encoder includes a downmixer for downmixing the multichannel signal to obtain a downmix signal, a linear prediction domain core encoder for encoding the downmix signal, wherein the downmix signal has a low band and a high band, wherein the linear prediction domain core encoder is configured to apply a bandwidth extension processing for parametrically encoding the high band, a filterbank for generating a spectral representation of the multichannel signal, and a joint multichannel encoder configured to process the spectral representation including the low band and the high band of the multichannel signal to generate multichannel information.

15. 发明申请

US20170309283A1 AUDIO ENCODER, AUDIO DECODER, METHODS FOR ENCODING AND DECODING AN AUDIO SIGNAL, AND A COMPUTER PROGRAM 审中-公开
公开(公告)号：US20170309283A1
公开(公告)日：2017-10-26
申请号：US15643908
申请日：2017-07-07
申请人： Fraunhofer-Gesellschaft zur Foerderung der Angewandten Forschung e.V.
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： G10L19/035 , G10L19/032 , G10L19/02 , G10L19/028 , G10L19/008 , G10L25/18
CPC分类号： G10L19/035 , G10L19/008 , G10L19/02 , G10L19/0204 , G10L19/028 , G10L19/032 , G10L25/18
摘要： An encoder for providing an audio stream on the basis of a transform-domain representation of an input audio signal includes a quantization error calculator configured to determine a multi-band quantization error over a plurality of frequency bands of the input audio signal for which separate band gain information is available. The encoder also includes an audio stream provider for providing the audio stream such that the audio stream includes information describing an audio content of the frequency bands and information describing the multi-band quantization error.A decoder for providing a decoded representation of an audio signal on the basis of an encoded audio stream representing spectral components of frequency bands of the audio signal includes a noise filler for introducing noise into spectral components of a plurality of frequency bands to which separate frequency band gain information is associated on the basis of a common multi-band noise intensity value.

16. 发明申请

US20160247516A1 ENCODER FOR ENCODING AN AUDIO SIGNAL, AUDIO TRANSMISSION SYSTEM AND METHOD FOR DETERMINING CORRECTION VALUES 有权
标题翻译：编码音频信号的编码器，音频传输系统和确定校正值的方法
公开(公告)号：US20160247516A1
公开(公告)日：2016-08-25
申请号：US15147844
申请日：2016-05-05
申请人： Fraunhofer-Gesellschaft zur Foerderung der angewandten Forschung e.V.
发明人： Konstantin SCHMIDT , Guillaume FUCHS , Matthias NEUSINGER , Martin DIETZ
IPC分类号： G10L19/16 , G10L19/038 , G10L19/06
CPC分类号： G10L19/167 , G10L19/038 , G10L19/06
摘要： An encoder for encoding an audio signal includes an analyzer for analyzing the audio signal and for determining analysis prediction coefficients from the audio signal. The encoder includes a converter for deriving converted prediction coefficients from the analysis prediction coefficients, a memory for storing a multitude of correction values and a calculator. The calculator includes a processor for processing the converted prediction coefficients to obtain spectral weighting factors. The calculator includes a combiner for combining the spectral weighting factors and the multitude of correction values to obtain corrected weighting factors. A quantizer of the calculator is configured for quantizing the converted prediction coefficients using the corrected weighting factors to obtain a quantized representation of the converted prediction coefficients. The encoder includes a bitstream former for forming an output signal based on the quantized representation of the converted prediction coefficients and based on the audio signal.
摘要翻译：用于编码音频信号的编码器包括用于分析音频信号并用于从音频信号确定分析预测系数的分析器。编码器包括用于从分析预测系数导出转换的预测系数的转换器，用于存储多个校正值的存储器和计算器。该计算器包括一个处理器，用于处理转换的预测系数以获得光谱加权因子。计算器包括用于组合频谱加权因子和多个校正值的组合器以获得校正的加权因子。计算器的量化器被配置为使用校正的加权因子对转换的预测系数进行量化，以获得经转换的预测系数的量化表示。编码器包括基于转换的预测系数的量化表示并基于音频信号形成输出信号的比特流形成器。

17. 发明申请

US20160027448A1 LOW-COMPLEXITY TONALITY-ADAPTIVE AUDIO SIGNAL QUANTIZATION 审中-公开
标题翻译：低复杂度的自适应音频信号量化
公开(公告)号：US20160027448A1
公开(公告)日：2016-01-28
申请号：US14812465
申请日：2015-07-29
申请人： Fraunhofer-Gesellschaft zur Foerderung der angewandten Forschung e.V.
发明人： Martin DIETZ , Guillaume FUCHS , Christian HELMRICH , Goran MARKOVIC
IPC分类号： G10L19/035 , G10H1/06 , G10L25/18 , G10L25/45 , G10L19/02 , G10L25/21
摘要： The invention provides an audio encoder for encoding an audio signal so as to produce therefrom an encoded signal, the audio encoder including: a framing device configured to extract frames from the audio signal; a quantizer configured to map spectral lines of a spectrum signal derived from the frame of the audio signal to quantization indices, wherein the quantizer has a dead-zone, in which the input spectral lines are mapped to quantization index zero; and a control device configured to modify the dead-zone; wherein the control device includes a tonality calculating device configured to calculate at least one tonality indicating value for at least one spectrum line or for at least one group of spectral lines, wherein the control device is configured to modify the dead-zone for the at least one spectrum line or the at least one group of spectrum lines depending on the respective tonality indicating value.
摘要翻译：本发明提供了一种音频编码器，用于对音频信号进行编码，从而产生编码信号，音频编码器包括：被配置为从音频信号中提取帧的成帧设备; 量化器，被配置为将从音频信号的帧导出的频谱信号的谱线映射到量化索引，其中量化器具有死区，其中输入谱线映射到量化索引为零; 以及配置为修改死区的控制装置; 其中所述控制装置包括音调计算装置，其被配置为计算至少一个频谱线或至少一组谱线的至少一个音调指示值，其中所述控制装置被配置为修改至少一个频谱线的死区取决于相应的音调指示值的一个频谱线或至少一组频谱线。

18. 发明申请

US20150364144A1 COMFORT NOISE ADDITION FOR MODELING BACKGROUND NOISE AT LOW BIT-RATES 审中-公开
标题翻译：舒适的噪声添加用于建模低比特率的背景噪音
公开(公告)号：US20150364144A1
公开(公告)日：2015-12-17
申请号：US14744788
申请日：2015-06-19
申请人： Fraunhofer-Gesellschaft zur Foerderung der angewandten Forschung e.V.
发明人： Guillaume FUCHS , Anthony LOMBARD , Emmanuel RAVELLI , Stefan DOEHLA , Jeremie LECOMTE , Martin DIETZ
IPC分类号： G10L19/012
CPC分类号： G10L19/012
摘要： The invention provides a decoder being configured for processing an encoded audio bitstream, wherein the decoder includes: a bitstream decoder configured to derive a decoded audio signal from the bitstream, wherein the decoded audio signal includes at least one decoded frame; a noise estimation device configured to produce a noise estimation signal containing an estimation of the level and/or the spectral shape of a noise in the decoded audio signal; a comfort noise generating device configured to derive a comfort noise signal from the noise estimation signal; and a combiner configured to combine the decoded frame of the decoded audio signal and the comfort noise signal in order to obtain an audio output signal.
摘要翻译：本发明提供了一种被配置用于处理编码音频比特流的解码器，其中解码器包括：比特流解码器，其被配置为从比特流导出解码的音频信号，其中解码的音频信号包括至少一个解码的帧; 噪声估计装置，被配置为产生包含解码音频信号中的噪声的电平和/或频谱形状的估计的噪声估计信号; 舒适噪声生成装置，被配置为从噪声估计信号导出舒适噪声信号; 以及组合器，其被配置为组合解码的音频信号的解码帧和舒适噪声信号，以便获得音频输出信号。

19. 发明申请

US20140081645A1 AUDIO ENCODER, AUDIO DECODER, METHOD FOR ENCODING AN AUDIO INFORMATION, METHOD FOR DECODING AN AUDIO INFORMATION AND COMPUTER PROGRAM USING A DETECTION OF A GROUP OF PREVIOUSLY-DECODED SPECTRAL VALUES 有权
标题翻译：音频编码器，音频解码器，用于编码音频信息的方法，用于解码音频信息的方法和使用检测以前解码的光谱值的计算机程序
公开(公告)号：US20140081645A1
公开(公告)日：2014-03-20
申请号：US14083412
申请日：2013-11-18
申请人： FRAUNHOFER-GESELLSCHAFT ZUR FOERDERUNG DER ANGEWANDTEN FORSCHUNG E.V.
发明人： Guillaume FUCHS , Vignesh SUBBARAMAN , Nikolaus RETTELBACH , Markus MULTRUS , Marc GAYER , Patrick WARMBOLD , Christian GRIEBEL , Oliver WEISS
IPC分类号： G10L19/008
CPC分类号： G10L19/008 , G10L19/0017 , G10L19/0208
摘要： An audio decoder for providing a decoded audio information includes a arithmetic decoder for providing a plurality of decoded spectral values on the basis of an arithmetically-encoded representation of the spectral values and a frequency-domain-to-time-domain converter for providing a time-domain audio representation using the decoded spectral values. The arithmetic decoder is configured to select a mapping rule describing a mapping of a code value onto a symbol code in dependence on a context state. The arithmetic decoder is configured to determine or modify the current context state in dependence on a plurality of previously-decoded spectral values. The arithmetic decoder is configured to detect a group of a plurality of previously-decoded spectral values, which fulfill, individually or taken together, a predetermined condition regarding their magnitudes, and to determine the current context state in dependence on a result of the detection.An audio encoder uses similar principles.
摘要翻译：用于提供经解码的音频信息的音频解码器包括算术解码器，用于根据频谱值的算术编码表示提供多个解码频谱值，以及频域 - 时域转换器，用于提供时间 - 使用解码的谱值的域音频表示。算术解码器被配置为根据上下文状态选择描述代码值到符号代码上的映射规则的映射规则。算术解码器被配置为根据多个先前解码的频谱值来确定或修改当前上下文状态。算术解码器被配置为检测多个先前解码的频谱值的组，它们单独地或一起实现关于其幅度的预定条件，并且根据检测结果来确定当前上下文状态。音频编码器使用类似的原理。

20. 发明申请

US20130332153A1 LINEAR PREDICTION BASED CODING SCHEME USING SPECTRAL DOMAIN NOISE SHAPING 有权
标题翻译：基于线性预测的编码方案，使用光谱域噪声形状
公开(公告)号：US20130332153A1
公开(公告)日：2013-12-12
申请号：US13966601
申请日：2013-08-14
申请人： Fraunhofer-Gesellschaft zur Foerderung der angewandten Forschung e.V.
发明人： Goran MARKOVIC , Guillaume FUCHS , Nikolaus RETTELBACH , Christian HELMRICH , Benjamin SCHUBERT
IPC分类号： G10L19/04
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 encoding concept which is linear prediction based and uses spectral domain noise shaping is rendered less complex at a comparable coding efficiency in terms of, for example, rate/distortion ratio, by using the spectral decomposition of the audio input signal into a spectrogram having a sequence of spectra for both linear prediction coefficient computation as well as spectral domain shaping based on the linear prediction coefficients. The coding efficiency may remain even if such a lapped transform is used for the spectral decomposition which causes aliasing and necessitates time aliasing cancellation such as critically sampled lapped transforms such as an MDCT.
摘要翻译：基于线性预测并且使用频谱噪声整形的编码概念在例如速率/失真比方面以相当的编码效率变得不那么复杂，通过使用音频输入信号的频谱分解成具有基于线性预测系数的线性预测系数计算以及频域整形的频谱序列。即使将这种重叠变换用于导致混叠的频谱分解并且需要时间混叠消除（例如诸如MDCT的批量采样的重叠变换）也可能保持编码效率。

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