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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

EP2720222A1 Apparatus and method for efficient synthesis of sinusoids and sweeps by employing spectral patterns 审中-公开
标题翻译：设备和方法，用于Sinosoiden和扫描的有效合成通过使用频谱图案
公开(公告)号：EP2720222A1
公开(公告)日：2014-04-16
申请号：EP12199266.3
申请日：2012-12-21
申请人： Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. , Friedrich-Alexander-Universität Erlangen-Nürnberg
发明人： Disch, Sascha , Sschubert, Benjamin , Geiger, Ralf , Edler, Bernd , Dietz, Martin
IPC分类号： G10L19/02
CPC分类号： G10L19/02 , G10L19/0212 , G10L19/032
摘要： An apparatus for generating an audio output signal based on an encoded audio signal spectrum is provided. The apparatus comprises a processing unit (115) for processing the encoded audio signal spectrum to obtain a decoded audio signal spectrum comprising a plurality of spectral coefficients, wherein each of the spectral coefficients has a spectral location within the encoded audio signal spectrum and a spectral value, wherein the spectral coefficients are sequentially ordered according to their spectral location within the encoded audio signal spectrum so that the spectral coefficients form a sequence of spectral coefficients. Moreover, the apparatus comprises a pseudo coefficients determiner (125) for determining one or more pseudo coefficients of the decoded audio signal spectrum, each of the pseudo coefficients having a spectral value. Furthermore, the apparatus comprises a replacement unit (135) for replacing at least one or more pseudo coefficients by a determined spectral pattern to obtain a modified audio signal spectrum, wherein the determined spectral pattern comprises at least two pattern coefficients, wherein each of the at least two pattern coefficients has a spectral value. Moreover, the apparatus comprises a spectrum-time-conversion unit (145) for converting the modified audio signal spectrum to a time-domain to obtain the audio output signal.
摘要翻译：提供了一种用于在在编码的音频信号的频谱基于音频输出信号产生的装置。该装置包括一个处理单元（115），用于处理经编码的音频信号的频谱，以获得解码音频信号的频谱，其包括频谱系数的多元worin每个频谱系数的具有经编码的音频信号频谱中的频谱位置和频谱值，worin频谱系数被顺序排列gemäß到他们的频谱位置中的编码的音频信号的频谱内所以做了频谱系数形成的频谱系数的序列。更上方，该装置包括一个伪系数确定器（125），用于确定的采矿一个或解码的音频信号的频谱的多个伪系数，每一个具有一个频谱值伪系数。进一步，该装置包括一个替换单元（135），用于通过确定性开采频谱图案替换至少一个或多个伪系数以获得经修改的音频信号的频谱，worin确定性开采频谱图案包括至少两个图案系数在worin每个至少两个图案系数具有光谱值。更上方，该装置包括一个频谱 - 时间变换单元（145）为修正的音频信号的频谱转换到时域，以获得所述音频输出信号。

4. 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.

5. 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.

6. 发明公开

EP3220389A1 ENCODER, DECODER AND METHOD FOR ENCODING AND DECODING AUDIO CONTENT USING PARAMETERS FOR ENHANCING A CONCEALMENT 有权
标题翻译：编码器，解码器和使用参数编码和解码音频内容以增强隐蔽性的方法
公开(公告)号：EP3220389A1
公开(公告)日：2017-09-20
申请号：EP17168268.5
申请日：2015-08-24
申请人： Fraunhofer Gesellschaft zur Förderung der angewandten Forschung e.V.
发明人： Lecomte, Jérémie , Schubert, Benjamin , Schnabel, Michael , Dietz, Martin
IPC分类号： G10L19/005 , G10L19/02 , G10L19/20 , G10L19/083 , G10L19/24
摘要： The present invention concerns an encoder for coding speech-like content and/or general audio content, wherein the encoder is configured to embed, at least in some frames, parameters in a bitstream, which parameters enhance a concealment in case an original frame is lost, corrupted or delayed, and a decoder for decoding speech-like content and/or general audio content, wherein the decoder is configured to use parameters which are sent later in time to enhance a concealment in case an original frame is lost, corrupted or delayed, as well as a method for encoding and a method for decoding.
摘要翻译：本发明涉及一种用于编码类话音内容和/或一般音频内容的编码器，其中编码器被配置为至少在一些帧中将比特流中的参数嵌入，该参数在原始帧丢失的情况下增强了隐藏，损坏或延迟的解码器，以及用于解码类语音内容和/或一般音频内容的解码器，其中解码器被配置为使用稍后发送的参数来增强隐藏，以防原始帧丢失，损坏或延迟以及编码方法和解码方法。

7. 发明授权

EP0911981B1 Verfahren und Vorrichtung zum Codieren bzw. Decodieren eines Audiosignals 有权
标题翻译：方法和编码装置或解码音频信号
公开(公告)号：EP0911981B1
公开(公告)日：2004-07-21
申请号：EP98119235.4
申请日：1998-10-12
申请人： Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
发明人： Sperschneider, Ralph , Dietz, Martin , Ehret, Andreas , Brandenburg, Karlheinz, Dr. , Gerhäuser, Heinz, Dr. , Ali Nowbakht-Irani , Pierre Lauber , Roland Bitto
IPC分类号： H03M7/40
CPC分类号： H04B1/662 , H03M7/40 , H03M13/35 , H04H20/46 , H04H20/71

8. 发明公开

EP0911981A2 Verfahren und Vorrichtung zum Codieren bzw. Decodieren eines Audiosignals 有权
标题翻译： Verfahren und Vorrichtung zum Codieren bzw Decodieren eines Audiosignals
公开(公告)号：EP0911981A2
公开(公告)日：1999-04-28
申请号：EP98119235.4
申请日：1998-10-12
申请人： FRAUNHOFER-GESELLSCHAFT ZUR FÖRDERUNG DER ANGEWANDTEN FORSCHUNG E.V.
发明人： Sperschneider, Ralph , Dietz, Martin , Ehret, Andreas , Brandenburg, Karlheinz, Dr. , Gerhäuser, Heinz, Dr.
IPC分类号： H03M7/30 , H04B1/66
CPC分类号： H04B1/662 , H03M7/40 , H03M13/35 , H04H20/46 , H04H20/71
摘要： Bei einem Verfahren zum Codieren eines Audiosignals, um einen codierten Bitstrom zu erhalten, werden zeitdiskrete Abtastwerte des Audiosignals in den Frequenzbereich transformiert, um Spektralwerte zu erhalten. Die Spektralwerte werden mit einer Codetabelle, die eine begrenzte Anzahl von Codewörtern unterschiedlicher Länge aufweist, codiert, um durch Codeworte codierte Spektralwerte zu erhalten, wobei die Länge eines einem Spektralwert zugeordneten Codeworts um so kürzer ist, je höher die Auftrittswahrscheinlichkeit des Spektralwerts ist. Anschließend wird ein Raster für den codierten Bitstrom festgelegt, wobei das Raster äquidistante Rasterpunkte aufweist, und wobei der Abstand der Rasterpunkte von der/den verwendeten Codetabelle/n abhängt. Um eine fehlerrobuste Huffman-Codierung zu erhalten, werden Prioritätscodewörter, die bestimmte Spektralwerte darstellen, die im Vergleich zu anderen Spektralwerten psychoakustisch bedeutsam sind, im Raster derart angeordnet, daß der Beginn jedes Prioritätscodeworts mit einem Rasterpunkt zusammenfällt.
摘要翻译：编码方法具有转换成频率范围的音频信号的离散采样值，以通过码表提供使用有限数量的不同长度的码字编码的频谱值。最常见的码字长度最短。提供了用于编码比特流的光栅，其包含在限定的光栅点（10,12,14）处的优先码字。独立地要求用于对音频信号进行编码的装置和用于解码音频信号的装置。

9. 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).

10. 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).

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