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1. 发明公开

EP3483884A1 SIGNAL FILTERING 审中-公开
公开(公告)号：EP3483884A1
公开(公告)日：2019-05-15
申请号：EP17201105.8
申请日：2017-11-10
申请人： Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. , Friedrich-Alexander-Universität Erlangen-Nürnberg
发明人： Ravelli, Emmanuel , Markovic, Goran , Dietz, Martin , Grill, Bernhard
IPC分类号： G10L19/26 , H03H17/02 , H04N19/117 , G10L19/04 , G10L19/00 , G10L19/09
摘要： There are discussed methods and systems for filtering an information input signal (11, 11a, x), divided into different update intervals, according to parameters varying with the update intervals, to obtain a filtered output signal (y, 15). A system (10) may comprise:
- a first filter unit (12) to filter a first filter input signal (11, x) at least at an initial subinterval ( T l ) in a current update interval ( T ) to obtain a first filter output signal (y', 13), according to parameters associated to the preceding update interval, the first filter unit (12) being configured to change the parameters along at least the initial subinterval ( T l ) from a higher-filtering status to a lower-filtering status; and
- a second filter unit (14) to filter a second filter input signal (13), at the initial interval ( T l ), according to parameters associated to the current update interval ( T ) to obtain a second filter output signal (15), the second filter unit (14) being configured to change the parameters along at least the initial subinterval ( T l ) from a lower-filtering status to a higher-filtering status.
The first filter input signal (11) is based on the information input signal (x), the first filter output signal (13) is an intermediate signal (y'), the second filter input signal is based on the intermediate signal (y'), and the filtered output signal (y) is based on the second filter output signal (15).

2. 发明公开

EP2980799A1 Apparatus and method for processing an audio signal using a harmonic post-filter 审中-公开
标题翻译：装置和方法用于使用谐波后滤波处理音频信号
公开(公告)号：EP2980799A1
公开(公告)日：2016-02-03
申请号：EP14178820.8
申请日：2014-07-28
申请人： Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. , Friedrich-Alexander-Universität Erlangen-Nürnberg
发明人： Ravelli, Emmanuel , Helmrich, Christian , Markovic, Goran , Neusinger, Matthias , Jander, Manuel , Dietz, Martin , Disch, Sascha Dr.-Ing.
IPC分类号： G10L19/26 , G10L21/02
CPC分类号： G10L19/26 , G10L19/0212 , G10L19/025 , G10L21/02
摘要： An apparatus for processing an audio signal having associated therewith a pitch lag information and a gain information, comprises a domain converter (100) for converting a first domain epresentation of the audio signal into a second domain representation of the audio signal; and a harmonic post-filter (104) for filtering the second domain representation of the audio signal, wherein the post-filter is based on a transfer function comprising a numerator and a denominator, wherein the numerator comprises a gain value indicated by the gain information, and wherein the denominator comprises an integer part of a pitch lag indicated by the pitch lag information and a multi-tap filter depending on a fractional part of the pitch lag.
摘要翻译：对于音频信号的处理的装置具有带间距为信息以及增益信息相关联的存在，包括域转换器（100），用于将音频信号的第一域ePresentation转换为音频信号的第二域表示; 和一个谐波后置滤波器（104），用于滤波所述音频信号的所述第二域表示，worin的后置滤波器是基于传递函数包括一个分子和分母，worin分子包括由增益信息所表示的增益值，分母worin包括是由于整数由俯仰指示的音高的部分是信息和多抽头滤波器取决于间距的小数部分是。

3. 发明公开

EP4447048A2 METHOD AND APPARATUS FOR PROCESSING AN AUDIO SIGNAL, COMPUTER READABLE MEDIUM AND AUDIO DECODER 审中-公开
公开(公告)号：EP4447048A2
公开(公告)日：2024-10-16
申请号：EP24198100.0
申请日：2015-07-03
申请人： Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
发明人： Ravelli, Emmanuel , Jander, Manuel , Pietrzyk, Grzegorz , Dietz, Martin , Gayer, Marc
IPC分类号： G11B27/038
CPC分类号： G11B27/038 , G10L21/0364 , G10L19/26 , G10L19/06
摘要： A method for processing an audio signal (100) includes receiving an encoded audio signal and generating a decoded audio signal by processing consecutive audio frames of the audio signal to avoid discontinuities. Processing consecutive audio frames of the audio signal to avoid discontinuities includes using linear predictive filtering for removing (S102, S104, S300-S308, S400-S402) a discontinuity (106a, 106b) between a filtered past frame and a filtered current frame of the audio signal. The method includes filtering the current frame of the audio signal and removing the discontinuity by modifying a beginning portion of the filtered current frame by a signal obtained by linear predictive filtering a predefined signal with initial states of the linear predictive filter defined on the basis of a last part of the unfiltered past frame filtered using the set of filter parameters for filtering the current frame.

4. 发明公开

EP4239634A1 AUDIO CODING USING A FREQUENCY DOMAIN PROCESSOR AND A TIME DOMAIN PROCESSOR 审中-实审
公开(公告)号：EP4239634A1
公开(公告)日：2023-09-06
申请号：EP23184408.5
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
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 (602) 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; an analyzer (604) for analyzing the frequency domain representation up to the maximum frequency to determine first spectral portions to be encoded with a first spectral resolution and second spectral regions to be encoded with a second spectral resolution, the second spectral resolution being lower than the first spectral resolution; a spectral encoder (606) for encoding the first spectral portions with the first spectral resolution and for encoding the second spectral portions with the second spectral resolution; a second encoding processor (610) for encoding a second different audio signal portion in the time domain; a controller (620) 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 (630) 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.

5. 发明授权

EP3396669B1 HARMONICITY-DEPENDENT CONTROLLING OF A HARMONIC FILTER TOOL 有权
公开(公告)号：EP3396669B1
公开(公告)日：2020-11-11
申请号：EP18177372.2
申请日：2015-07-27
申请人： FRAUNHOFER-GESELLSCHAFT zur Förderung der angewandten Forschung e.V.
发明人： Markovic, Goran , Helmrich, Christian , Ravelli, Emmanuel , Jander, Manuel , Döhla, Stefan
IPC分类号： G10L19/26 , G10L19/025 , G10L19/028 , G10L19/12 , G10L19/22 , G10L25/90 , G10L25/21

6. 发明公开

EP2980797A1 Audio decoder, method and computer program using a zero-input-response to obtain a smooth transition 审中-公开
标题翻译：音频解码器，方法和计算机程序的零输入响应，以产生平滑的过渡
公开(公告)号：EP2980797A1
公开(公告)日：2016-02-03
申请号：EP14178830.7
申请日：2014-07-28
申请人： Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
发明人： Ravelli, Emmanuel , Fuchs, Guillaume , Disch, Sascha , Multrus, Markus , Pietrzyk, Grzegorz , Schubert, Benjamin
IPC分类号： G10L19/20
CPC分类号： G10L19/20 , G10L19/02 , G10L19/12
摘要： An audio decoder (100;200;300) for providing a decoded audio information (112;212;312) on the basis of an encoded audio information (110;210;310), the audio decoder comprises a linear-prediction-domain decoder (120;220;320) configured to provide a first decoded audio information (122;222;322; S C (n)) on the basis of an audio frame encoded in a linear prediction domain, a frequency domain decoder (130;230;330) configured to provide a second decoded audio information (132;232;332; S M (n)) on the basis of an audio frame encoded in a frequency domain, and a transition processor (140;240;340).The transition processor is configured to obtain a zero-input-response (150; 256;348) of a linear predictive filtering(148; 254; 346), wherein an initial state (146;252;344) of the linear predictive filtering is defined in dependence on the first decoded audio information and the second decoded audio information. The transition processor is also configured to modify the second decoded audio information (132; 232;332;S M (n)), which is provided on the basis of an audio frame encoded in the frequency domain following an audio frame encoded in the linear prediction domain, in dependence on the zero-input-response, to obtain a smooth transition between the first decoded audio information (S C (n)) and the modified second decoded audio information ( S M ^ n ).
摘要翻译：音频解码器（100; 200; 300），用于提供已解码音频信息的编码音频信息的基础上（112; 312; 212）（110; 210; 310），所述音频解码器包括线性预测域解码器（120; 220; 320）被配置为提供一个第一解码的音频信息（122; SC（N）222; 322）; 230线性预测域，频域解码器（130编码的音频帧的基础上; 330）被配置为提供第二经解码的音频信息（132; 232; 332; SM（N））在频域编码的音频帧，以及过渡处理器（140的基础上; 240; 340）。该过渡处理器被配置为获得一个零输入响应（; 254; 346 148），worin到初始化状态的线性预测滤波（150; 348; 256）（146; 252; 344）的线性预测滤波的依赖被定义在第一解码的音频信息和第二解码的音频信息。过渡处理器被如此配置以修改所述第二解码的音频信息（132; 232; 332; SM（N）），所有这些都设置在频域中编码的音频帧的基础上继在所述线性预测编码的音频帧域，在该零输入响应的依赖性，以获得第一解码的音频信息之间和所述修改后的第二解码的音频信息（SM ^ N）的平滑过渡（（SC n））的。

7. 发明公开

EP3806094A1 CONCEPT FOR ENCODING AN AUDIO SIGNAL AND DECODING AN AUDIO SIGNAL USING SPEECH RELATED SPECTRAL SHAPING INFORMATION 审中-公开
公开(公告)号：EP3806094A1
公开(公告)日：2021-04-14
申请号：EP20210767.8
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
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.

8. 发明公开

EP3503097A3 APPARATUS AND METHOD FOR ENCODING OR DECODING A MULTI-CHANNEL SIGNAL USING SPECTRAL-DOMAIN RESAMPLING 有权转让
公开(公告)号：EP3503097A3
公开(公告)日：2019-07-03
申请号：EP19157001.9
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 , Jaegers, Wolfgang
IPC分类号： G10L19/008 , G10L19/02 , G10L19/022
摘要： An apparatus for encoding a multi-channel signal comprising at least two channels, comprises: a time-spectral converter (1000) 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 (1211) being related to the input sampling rate; a multi-channel processor (1010) for applying a joint multi-channel processing to the sequences of blocks of spectral values or to resampled sequences of blocks of spectral values to obtain at least one result sequence of blocks of spectral values comprising information related to the at least two channels; a spectral domain resampler (1020) for resampling the blocks of the result sequences in the frequency domain or for resampling the sequences of blocks of spectral values for the at least two channels in the frequency domain to obtain a resampled sequence of blocks of spectral values, wherein a block of the resampled sequence of blocks of spectral values 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 of spectral values into a time domain representation or for converting the result sequence of blocks of spectral values into a time domain representation comprising an output sequence of blocks of sampling values having associated an output sampling rate being different from the input sampling rate; and a core encoder (1040) for encoding the output sequence of blocks of sampling values to obtain an encoded multi-channel signal (1510).

9. 发明公开

EP3483886A1 SELECTING PITCH LAG 审中-公开
公开(公告)号：EP3483886A1
公开(公告)日：2019-05-15
申请号：EP17201091.0
申请日：2017-11-10
申请人： Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
发明人： Ravelli, Emmanuel , Dietz, Martin , Schnabel, Michael , Tritthart, Arthur , TSCHEKALINSKIJ, Alexander
IPC分类号： G10L25/90 , G10L25/06 , G10L19/26 , G10L19/005
摘要： There are provided methods and apparatus capable of performing a low complexity pitch detection procedure, e.g., for long term postfiltering, LTPF, encoding.
One inventive apparatus (10, 60a, 110) for encoding an information signal including a plurality of frames, comprises:
a first estimator (11) configured to obtain a first estimate (14, T 1 ), the first estimate being an estimate of a pitch lag for a current frame (13);
a second estimator (12) configured to obtain a second estimate (16, T 2 ), the second estimate being another estimate of a pitch lag for the current frame (13),
a selector (17) configured to choose (S103) a selected value (19, T best ) by performing a selection between the first estimate (14, T 1 ) and the second estimate (16, T 2 ) on the basis of at least one correlation measurement (23, 25),
wherein the second estimator (12) is conditioned by the pitch lag (51, 19") selected at the previous frame so as to obtain the second estimate (16, T 2 ) for the current frame (13).

10. 发明公开

EP2980794A1 Audio encoder and decoder using a frequency domain processor and a time domain processor 审中-公开
标题翻译： Audiocodierer und -Decodierer mit einem常见问题解答
公开(公告)号：EP2980794A1
公开(公告)日：2016-02-03
申请号：EP14178817.4
申请日：2014-07-28
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： G10L19/18 , G10L19/02 , G10L19/028 , G10L19/032 , G10L19/04 , G10L19/06 , G10L19/24 , G10L19/265 , G10L21/038
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 (602) 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; an analyzer (604) for analyzing the frequency domain representation up to the maximum frequency to determine first spectral portions to be encoded with a first spectral resolution and second spectral regions to be encoded with a second spectral resolution, the second spectral resolution being lower than the first spectral resolution; a spectral encoder (606) for encoding the first spectral portions with the first spectral resolution and for encoding the second spectral portions with the second spectral resolution; a second encoding processor (610) for encoding a second different audio signal portion in the time domain; a controller (620) 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 (630) 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：一种用于对音频信号进行编码的音频编码器，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），其中所述第一编码处理器（600）包括：时间频率转换器（602）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; 分析器（604），用于分析直到最大频率的频域表示，以确定要用第一光谱分辨率编码的第一光谱部分和要用第二光谱分辨率编码的第二光谱区域，第二光谱分辨率低于第一光谱分辨率; 用于对具有第一光谱分辨率的第一光谱部分进行编码并用第二光谱分辨率对第二光谱部分进行编码的光谱编码器（606）用于对时域中的第二不同音频信号部分进行编码的第二编码处理器（610）配置用于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的控制器（620），以及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（630），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。

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