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1. 发明申请

WO2020165265A1 DECODER AND DECODING METHOD FOR LC3 CONCEALMENT INCLUDING FULL FRAME LOSS CONCEALMENT AND PARTIAL FRAME LOSS CONCEALMENT 审中-公开
公开(公告)号：WO2020165265A1
公开(公告)日：2020-08-20
申请号：PCT/EP2020/053620
申请日：2020-02-12
申请人： FRAUNHOFER-GESELLSCHAFT ZUR FÖRDERUNG DER ANGEWANDTEN FORSCHUNG E.V.
发明人： TOMASEK, Adrian , SPERSCHNEIDER, Ralph , BÜTHE, Jan , BENNDORF, Conrad , DIETZ, Martin , SCHNELL, Markus , SCHLEGEL, Maximilian
IPC分类号： G10L19/005 , H04L1/00 , G10L19/02
摘要： Fig. 1 illustrates a decoder (100) for decoding a current frame to reconstruct an audio signal according to an embodiment. The audio signal is encoded within the current frame. The current frame comprises a current bitstream payload. The current bitstream payload comprises a plurality of payload bits. The plurality of payload bits encodes a plurality of spectral lines of a spectrum of the audio signal. Each of the payload bits exhibits a position within the current bitstream payload. The decoder (100) comprises a decoding module (110) and an output interface (120). The decoding module (110) is configured to reconstruct the audio signal. The output interface (120) is configured to output the audio signal. The decoding module (110) comprises an error concealment mode, wherein, if the decoding module (110) is in said error concealment mode, the decoding module (110) is configured to reconstruct the audio signal by conducting error concealment for those spectral lines of the spectrum of the audio signal, which exhibit a frequency being greater than a threshold frequency. And/or, if error concealment is conducted by the decoding module (110), the decoding module (110) is configured to conduct error concealment in a way that depends on whether or not a previous bitstream payload of a previous frame preceding the current frame encodes a signal component of the audio signal which is tonal or harmonic.

2. 发明申请

WO2017125563A1 APPARATUS AND METHOD FOR ESTIMATING AN INTER-CHANNEL TIME DIFFERENCE 审中-公开
标题翻译：用于估计信道间时间差的装置和方法
公开(公告)号：WO2017125563A1
公开(公告)日：2017-07-27
申请号：PCT/EP2017/051214
申请日：2017-01-20
申请人： FRAUNHOFER-GESELLSCHAFT ZUR FÖRDERUNG DER ANGEWANDTEN FORSCHUNG E.V.
发明人： BAYER, Stefan , FOTOPOULOU, Eleni , MULTRUS, Markus , FUCHS, Guillaume , RAVELLI, Emmanuel , SCHNELL, Markus , DÖHLA, Stefan , JÄGERS, Wolfgang , DIETZ, Martin , MARKOVIC, Goran
IPC分类号： G10L19/008 , G10L19/02
CPC分类号： G10L19/008 , G10L19/02 , G10L19/022 , G10L19/04 , H04S2420/03
摘要： An apparatus for estimating an inter-channel time difference between a first channel signal and a second channel signal, comprises: a calculator (1020) 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 (1010) for estimating a characteristic of a spectrum of the first channel signal or the second channel signal for the time block; a smoothing filter (1030) for smoothing the cross-correlation spectrum over time using the spectral characteristic to obtain a smoothed cross- correlation spectrum; and a processor (1040) for processing the smoothed cross- correlation spectrum to obtain the inter-channel time difference.
摘要翻译：一种用于估计第一信道信号和第二信道信号之间的信道间时间差的装置，包括：计算器（1020），用于计算来自第一信道信号和第二信道信号的时间块的互相关谱时间块中的信道信号和时间块中的第二信道信号; 谱特性估计器（1010），用于估计所述时间块的所述第一信道信号或所述第二信道信号的谱的特性; 平滑滤波器（1030），用于使用所述频谱特性随时间平滑所述互相关频谱以获得经平滑的互相关频谱; 和用于处理平滑的互相关频谱以获得信道间时间差的处理器（1040）。

3. 发明申请

WO2017125562A1 APPARATUSES AND METHODS FOR ENCODING OR DECODING A MULTI-CHANNEL AUDIO SIGNAL USING FRAME CONTROL SYNCHRONIZATION 审中-公开
标题翻译：使用帧控制同步对多通道音频信号进行编码或解码的装置和方法
公开(公告)号：WO2017125562A1
公开(公告)日：2017-07-27
申请号：PCT/EP2017/051212
申请日：2017-01-20
申请人： FRAUNHOFER-GESELLSCHAFT ZUR FÖRDERUNG DER ANGEWANDTEN FORSCHUNG E.V.
发明人： FUCHS, Guillaume , RAVELLI, Emmanuel , MULTRUS, Markus , SCHNELL, Markus , DÖHLA, Stefan , DIETZ, Martin , MARKOVIC, Goran , FOTOPOULOU, Eleni , BAYER, Stefan , JÄGERS, Wolfgang
IPC分类号： G10L19/022 , G10L19/008
CPC分类号： G10L19/008 , G10L19/02 , G10L19/022 , G10L19/04 , H04S2420/03
摘要： Encoding a multi-channel audio signal using a time-spectral converter for converting sequences of blocks of sampling values into sequences of blocks of spectral values; a multi-channel processor for applying a joint multi-channel processing to blocks of spectral values to obtain at least one result sequence of blocks; a spectral-time converter for converting the result sequence of blocks of spectral values into a time domain representation comprising an output sequence of blocks of sampling values; and a core encoder for encoding the output sequence of blocks of sampling values to obtain an encoded multi-channel signal, wherein the core encoder operates with a first frame control, and wherein the time-spectral converter or the spectral-time converter operate with a second frame control being synchronized to the first frame control, wherein the start or end frame border is in a predetermined relation to a start or end instant of an overlapping portion of a window used by the time-spectral converter.
摘要翻译：使用时间频谱转换器编码多声道音频信号，用于将采样值块的序列转换为频谱值的块序列; 多信道处理器，用于对频谱值块应用联合多信道处理以获得块的至少一个结果序列; 频谱时间转换器，用于将频谱值块的结果序列转换为时域表示，该时域表示包括采样值块的输出序列; 以及核心编码器，用于对采样值的输出序列进行编码以获得编码的多声道信号，其中核心编码器以第一帧控制进行操作，并且其中时间谱转换器或频谱时间转换器以第二帧控制与第一帧控制同步，其中开始或结束帧边界与由时间 - 频谱转换器使用的窗口的重叠部分的开始或结束时刻成预定关系。

4. 发明申请

WO2017125559A1 APPARATUSES AND METHODS FOR ENCODING OR DECODING AN AUDIO MULTI-CHANNEL SIGNAL USING SPECTRAL-DOMAIN RESAMPLING 审中-公开
标题翻译：使用谱域重构对音频多通道信号进行编码或解码的设备和方法
公开(公告)号：WO2017125559A1
公开(公告)日：2017-07-27
申请号：PCT/EP2017/051208
申请日：2017-01-20
申请人： FRAUNHOFER-GESELLSCHAFT ZUR FÖRDERUNG DER ANGEWANDTEN FORSCHUNG E.V.
发明人： FUCHS, Guillaume , RAVELLI, Emmanuel , MULTRUS, Markus , SCHNELL, Markus , DÖHLA, Stefan , DIETZ, Martin , MARKOVIC, Goran , FOTOPOULOU, Eleni , BAYER, Stefan , JÄGERS, Wolfgang
IPC分类号： G10L19/008 , G10L19/02
CPC分类号： G10L19/008 , G10L19/02 , G10L19/022 , G10L19/04 , H04S2420/03
摘要： An apparatus converting sequences of blocks of sample values of at least two channels into sequences of blocks of spectral values, wherein a block of sampling values has an input sampling rate, a multi-channel processor (1010) for applying a joint multi-channel processing to the sequences of blocks or to resampled sequences of blocks to obtain at least one result sequence of blocks of spectral values; a spectral domain resampler (1020) for resampling the blocks of the result sequences or for resampling the sequences of blocks of spectral values to obtain a resampled sequence of blocks of spectral values, wherein a block of the resampled sequence of blocks has spectral values up to a maximum output frequency (1231, 1221) being different from the maximum input frequency (1211); a spectral-time converter for converting the resampled sequence of blocks or the result sequence of blocks into a time domain; and a core encoder (1040) for encoding the output sequence of blocks.
摘要翻译：一种将至少两个信道的采样值的块的序列转换成频谱值的块的序列的设备，其中采样值块具有输入采样率，多信道处理器（1010）用于将联合多信道处理应用于块序列或块重采样序列以获得频谱值块的至少一个结果序列; 频谱域重采样器（1020），用于重新采样结果序列的块或用于重新采样频谱值的块的序列以获得频谱值的块的重采样序列，其中重新采样的块序列的块具有高达最大输出频率（1231,1221）与最大输入频率（1211）不同; 频谱时间转换器，用于将重新采样的块的序列或块的结果序列转换为时域; 和用于编码块的输出序列的核心编码器（1040）。

5. 发明申请

WO2016016124A1 AUDIO ENCODER AND DECODER USING A FREQUENCY DOMAIN PROCESSOR, A TIME DOMAIN PROCESSOR, AND A CROSS PROCESSOR FOR CONTINUOUS INITIALIZATION 审中-公开
标题翻译：音频编码器和解码器使用频域处理器，时域处理器和连续初始化的交叉处理器
公开(公告)号：WO2016016124A1
公开(公告)日：2016-02-04
申请号：PCT/EP2015/067005
申请日：2015-07-24
申请人： FRAUNHOFER-GESELLSCHAFT ZUR FÖRDERUNG DER ANGEWANDTEN FORSCHUNG E.V.
发明人： DISCH, Sascha , DIETZ, Martin , MULTRUS, Markus , FUCHS, Guillaume , RAVELLI, Emmanuel , NEUSINGER, Matthias , SCHNELL, Markus , SCHUBERT, Benjamin , GRILL, Bernhard
IPC分类号： G10L19/18 , G10L19/02 , G10L19/028 , G10L19/04 , G10L19/24 , G10L21/038
CPC分类号： G10L19/0208 , G10L19/02 , G10L19/022 , G10L19/028 , G10L19/04 , G10L19/083 , G10L19/18 , G10L19/24 , G10L19/26 , G10L21/038 , G10L2019/0001
摘要： An audio encoder for encoding an audio signal, comprises: a first encoding processor (600) for encoding a first audio signal portion in a frequency domain, wherein the first encoding processor (600) comprises: a time frequency converter for converting the first audio signal portion into a frequency domain representation having spectral lines up to a maximum frequency of the first audio signal portion; a spectral encoder for encoding the frequency domain representation; a second encoding processor for encoding a second different audio signal portion in the time domain; a cross-processor (700) for calculating, from the encoded spectral representation of the first audio signal portion, initialization data of the second encoding processor (610), so that the second encoding processing (610) is initialized to encode the second audio signal portion immediately following the first audio signal portion in time in the audio signal; a controller configured for analyzing the audio signal and for determining, which portion of the audio signal is the first audio signal portion encoded in the frequency domain and which portion of the audio signal is the second audio signal portion encoded in the time domain; and an encoded signal former for forming an encoded audio signal comprising a first encoded signal portion for the first audio signal portion and a second encoded signal portion for the second audio signal portion.
摘要翻译：一种用于编码音频信号的音频编码器，包括：用于对频域中的第一音频信号部分进行编码的第一编码处理器（600），其中第一编码处理器（600）包括：时间频率转换器，用于将第一音频信号部分转换成具有高达第一音频信号部分的最大频率的频谱线的频域表示; 用于对频域表示进行编码的频谱编码器; 第二编码处理器，用于对时域中的第二不同音频信号部分进行编码; 用于从所述第一音频信号部分的经编码的频谱表示计算所述第二编码处理器（610）的初始化数据的交叉处理器（700），使得所述第二编码处理（610）被初始化以对所述第二音频信号在音频信号中紧接在第一音频信号部分之后的部分; 控制器，被配置为分析音频信号并确定音频信号的哪个部分是在频域中编码的第一音频信号部分，并且音频信号的哪个部分是在时域中编码的第二音频信号部分; 以及用于形成编码音频信号的编码信号形成器，包括用于第一音频信号部分的第一编码信号部分和用于第二音频信号部分的第二编码信号部分。

6. 发明申请

WO2015055531A1 CONCEPT FOR ENCODING AN AUDIO SIGNAL AND DECODING AN AUDIO SIGNAL USING SPEECH RELATED SPECTRAL SHAPING INFORMATION 审中-公开
标题翻译：使用语音相关光谱形状信息编码音频信号和解码音频信号的概念
公开(公告)号：WO2015055531A1
公开(公告)日：2015-04-23
申请号：PCT/EP2014/071767
申请日：2014-10-10
申请人： FRAUNHOFER-GESELLSCHAFT ZUR FÖRDERUNG DER ANGEWANDTEN FORSCHUNG E.V.
发明人： FUCHS, Guillaume , MULTRUS, Markus , RAVELLI, Emmanuel , SCHNELL, Markus
IPC分类号： G10L19/20 , G10L19/083
CPC分类号： G10L19/12 , G10L19/032 , G10L19/083 , G10L19/20 , G10L25/93
摘要： According to an aspect of the present invention an encoder for encoding an audio signal comprises an analyzer configured for deriving prediction coefficients and a residual signal from a frame of the audio signal. The encoder comprises a formant information calculator configured for calculating a speech related spectral shaping information from the prediction coefficients, a gain parameter calculator configured for calculating a gain parameter from an unvoiced residual signal and the spectral shaping information and a bitstream former configured for forming an output signal based on an information related to a voiced signal frame, the gain parameter or a quantized gain parameter and the prediction coefficients.
摘要翻译：根据本发明的一个方面，用于对音频信号进行编码的编码器包括被配置为从音频信号的帧导出预测系数和残差信号的分析器。所述编码器包括：共振峰信息计算器，被配置为从所述预测系数计算与语音有关的频谱整形信息;增益参数计算器，被配置为从无声残差信号和所述频谱整形信息计算增益参数;以及比特流形成器，被配置用于形成输出基于与有声信号帧相关的信息，增益参数或量化增益参数以及预测系数的信号。

7. 发明申请

WO2011048118A1 AUDIO SIGNAL ENCODER, AUDIO SIGNAL DECODER, METHOD FOR PROVIDING AN ENCODED REPRESENTATION OF AN AUDIO CONTENT, METHOD FOR PROVIDING A DECODED REPRESENTATION OF AN AUDIO CONTENT AND COMPUTER PROGRAM FOR USE IN LOW DELAY APPLICATIONS 审中-公开
标题翻译：音频信号编码器，音频信号解码器，用于提供音频内容的编码表示的方法，用于提供音频内容的解码表示的方法和用于低延迟应用的计算机程序
公开(公告)号：WO2011048118A1
公开(公告)日：2011-04-28
申请号：PCT/EP2010/065753
申请日：2010-10-19
申请人： FRAUNHOFER GESELLSCHAFT ZUR FÖRDERUNG DER ANGEWANDTEN FORSCHUNG E.V. , GEIGER, Ralf , SCHNELL, Markus , LECOMTE, Jérémie , SCHMIDT, Konstantin , FUCHS, Guillaume , RETTELBACH, Nikolaus
发明人： GEIGER, Ralf , SCHNELL, Markus , LECOMTE, Jérémie , SCHMIDT, Konstantin , FUCHS, Guillaume , RETTELBACH, Nikolaus
IPC分类号： G10L19/02 , G10L19/04 , G10L19/14
CPC分类号： G10L19/0212 , G10L19/022 , G10L19/20
摘要： An audio signal encoder (100) comprises a transform-domain path (12) configured to obtain a set of spectral coefficients (124) and noise-shaping information (126) on the basis of a time-domain representation (122) of a portion of the audio content to be encoded in a transform-domain mode. The transform-domain path comprises a time-domain-to-frequency-domain converter (130) configured to window a time-domain representation of the audio content, or a pre-processed version thereof, to obtain a windowed representation of the audio content, and to apply a time-domain-to-frequency-domain conversion, to derive a set of spectral coefficients from the windowed time-domain representation of the audio content. The audio signal decoder comprises a CELP path (140) configured to obtain an code-excitation information (144) and a linear-prediction-domain parameter information (146) on the basis of a portion of the audio content to be encoded in a CELP mode. The time-domain-to-frequency-domain converter (136) is configured to apply a predetermined asymmetric analysis window (520) for a windowing of a current portion of the audio content to be encoded in the transform-domain mode and following a portion of the audio content encoded in the transform-domain mode both if the current portion of the audio content is followed by a subsequent portion of the audio content to be encoded in the transform-domain mode and if the current portion of the audio content is followed by a subsequent portion of the audio content to be encoded in the CELP mode. The audio signal encoder is configured to selectively provide an aliasing cancellation information (164) if the current portion of the audio content is followed by a subsequent portion of the audio content to be encoded in the CELP mode.
摘要翻译：音频信号编码器（100）包括变换域路径（12），其经配置以基于部分的时域表示（122）获得一组频谱系数（124）和噪声整形信息（126）要以变换域模式编码的音频内容。变换域路径包括被配置为对音频内容的时域表示或其预处理版本进行窗口化的时域 - 频域转换器（130），以获得音频内容的窗口化表示并且应用时域到频域转换，从音频内容的加窗时域表示中导出一组频谱系数。音频信号解码器包括CELP路径（140），其被配置为基于要在CELP中编码的音频内容的一部分来获得码激励信息（144）和线性预测域参数信息（146）模式。时域 - 频域转换器（136）被配置为对预定的非对称分析窗口（520）应用于在变换域模式中对要编码的音频内容的当前部分的加窗，并且遵循一部分如果音频内容的当前部分后面是要以变换域模式编码的音频内容的后续部分，并且如果跟随音频内容的当前部分，则以变换域模式编码的音频内容通过CELP模式中要编码的音频内容的后续部分。音频信号编码器被配置为如果音频内容的当前部分跟随着要以CELP模式编码的音频内容的后续部分，则选择性地提供混叠消除信息（164）。

8. 发明申请

WO2009109374A2 MIXING OF INPUT DATA STREAMS AND GENERATION OF AN OUTPUT DATA STREAM THEREFROM 审中-公开
标题翻译：输入数据流的混合和其输出数据流的生成
公开(公告)号：WO2009109374A2
公开(公告)日：2009-09-11
申请号：PCT/EP2009/001534
申请日：2009-03-04
申请人： FRAUNHOFER-GESELLSCHAFT ZUR FÖRDERUNG DER ANGEWANDTEN FORSCHUNG E.V. , SCHNELL, Markus , LUTZKY, Manfred , MULTRUS, Markus
发明人： SCHNELL, Markus , LUTZKY, Manfred , MULTRUS, Markus
IPC分类号： G10L19/00
CPC分类号： G10L21/038 , G10L19/008 , G10L19/265
摘要： An apparatus (500) for mixing a plurality of input data streams (510) is described, wherein the input data streams (510) each comprise a frame (540) of audio data in the spectral domain, a frame (540) of an input data stream (510) comprising spectral information for a plurality of spectral components. The apparatus comprises a processing unit (520) adapted to compare the frames (540) of the plurality of input data streams (510). The processing unit (520) is further adapted to determine, based on the comparison, for a spectral component of an output frame (550) of an output data stream (530), exactly one input data stream (510) of the plurality of input data streams (510). The processing unit (520) is further adapted to generate the output data stream (530) by copying at least a part of an information of a corresponding spectral component of the frame of the determined data stream (510) to describe the spectral component of the output frame (550) of the output data stream (530). Further or alternatively, the control value of the frames (540) of the first input data stream (510-1) and the second input data stream (510-2) may be compared to yield a comparison result and, if the comparison result is positive, the output data stream (530) comprising an output frame (550) may be generated such that the output frame (550) comprises a control value equal to that of the first and second input data streams (510) and payload data derived from the payload data of the frames of the first and second input data streams by processing the audio data in the spectral domain.
摘要翻译：描述了用于混合多个输入数据流（510）的装置（500），其中输入数据流（510）各自包括频谱域中的音频数据的帧（540），输入的帧（540）数据流（510）包括用于多个频谱分量的频谱信息。该装置包括适于比较多个输入数据流（510）的帧（540）的处理单元（520）。处理单元（520）还适于基于比较确定对于输出数据流（530）的输出帧（550）的频谱分量，多个输入的恰好一个输入数据流（510）数据流（510）。处理单元（520）还适于通过复制所确定的数据流（510）的帧的对应频谱分量的信息的至少一部分来生成输出数据流（530），以描述所确定的数据流的频谱分量输出数据流（530）的输出帧（550）。进一步或替代地，可以比较第一输入数据流（510-1）和第二输入数据流（510-2）的帧（540）的控制值以产生比较结果，并且如果比较结果为可以生成包括输出帧（550）的输出数据流（530），使得输出帧（550）包括等于第一和第二输入数据流（510）的控制值和从通过处理频谱域中的音频数据来输出第一和第二输入数据流的帧的有效载荷数据。

9. 发明申请

WO2020164753A1 DECODER AND DECODING METHOD SELECTING AN ERROR CONCEALMENT MODE, AND ENCODER AND ENCODING METHOD 审中-公开
公开(公告)号：WO2020164753A1
公开(公告)日：2020-08-20
申请号：PCT/EP2019/065209
申请日：2019-06-11
申请人： FRAUNHOFER-GESELLSCHAFT ZUR FÖRDERUNG DER ANGEWANDTEN FORSCHUNG E.V.
发明人： TOMASEK, Adrian , SPERSCHNEIDER, Ralph , BUETHE, Jan , BENNDORF, Conrad , DIETZ, Martin , SCHNELL, Markus , SCHLEGEL, Maximilian
IPC分类号： H04L1/00 , G10L19/005
摘要： A decoder (100) for decoding a frame to reconstruct a signal portion of a signal is provided. The signal portion is encoded within the frame, wherein the frame comprises a bitstream payload and two or more redundancy bits, wherein the bitstream payload comprises a plurality of payload bits, wherein each of the payload bits exhibits a position within the bitstream payload. The decoder (100) comprises a channel decoding module (110), being configured to detect, depending on the two or more redundancy bits, whether the bitstream payload comprises one or more corrupted bits being one or more of the payload bits that are distorted or that are likely to be distorted. Moreover, the decoder (100) comprises a source decoding module (120). If the channel decoding module (110) has not detected any corrupted bits within the bitstream payload, the source decoding module (120) is configured to decode the bitstream payload without conducting error concealment to reconstruct the signal portion. If the channel decoding module (110) has detected the one or more corrupted bits within the bitstream payload, the source decoding module (120) is configured to select a selected error concealment mode of two or more error concealment modes depending on the position of at least one of the one or more corrupted bits within the bitstream payload and depending on a signal characteristic of the signal portion of the signal, and is configured to conduct error concealment depending on the selected error concealment mode to reconstruct the signal portion.

10. 发明申请

WO2020164751A1 DECODER AND DECODING METHOD FOR LC3 CONCEALMENT INCLUDING FULL FRAME LOSS CONCEALMENT AND PARTIAL FRAME LOSS CONCEALMENT 审中-公开
公开(公告)号：WO2020164751A1
公开(公告)日：2020-08-20
申请号：PCT/EP2019/065172
申请日：2019-06-11
申请人： FRAUNHOFER-GESELLSCHAFT ZUR FÖRDERUNG DER ANGEWANDTEN FORSCHUNG E.V.
发明人： TOMASEK, Adrian , SPERSCHNEIDER, Ralph , BÜTHE, Jan , BENNDORF, Conrad , DIETZ, Martin , SCHNELL, Markus , SCHLEGEL, Maximilian
IPC分类号： G10L19/005 , H04L1/00 , G10L19/02
摘要： Fig. 1 illustrates a decoder (100) for decoding a current frame to reconstruct an audio signal according to an embodiment. The audio signal is encoded within the current frame. The current frame comprises a current bitstream payload. The current bitstream payload comprises a plurality of payload bits. The plurality of payload bits encodes a plurality of spectral lines of a spectrum of the audio signal. Each of the payload bits exhibits a position within the current bitstream payload. The decoder (100) comprises a decoding module (110) and an output interface (120). The decoding module (110) is configured to reconstruct the audio signal. The output interface (120) is configured to output the audio signal. The decoding module (110) comprises an error concealment mode, wherein, if the decoding module (110) is in said error concealment mode, the decoding module (110) is configured to reconstruct the audio signal by conducting error concealment for those spectral lines of the spectrum of the audio signal, which exhibit a frequency being greater than a threshold frequency. And/or, if error concealment is conducted by the decoding module (110), the decoding module (110) is configured to conduct error concealment in a way that depends on whether or not a previous bitstream payload of a previous frame preceding the current frame encodes a signal component of the audio signal which is tonal or harmonic.

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