专利快速检索-快速检索全球专利，免费商用专利数据库-IPRDB

1. 发明专利

BR112015007137B1 aparelho para codificar um sinal de fala que emprega acelp no domínio de autocorrelação 未知
公开(公告)号：BR112015007137B1
公开(公告)日：2021-07-13
申请号：BR112015007137
申请日：2013-07-31
申请人： FRAUNHOFER GES FORSCHUNG
发明人： BACKSTROM TOM , DIETZ MARTIN , FUCHS GUILLAUME , HELMRICH CHRISTIAN , MULTRUS MARKUS
IPC分类号： G10L19/107
摘要： aparelho para codificar um sinal de fala que emprega acelp no domínio de autocorrelação. um aparelho para codificar um sinal de fala pela determinação de um vetor do livro de códigos de um algoritmo de codificação de fala é fornecido. o aparelho compreende um determinador de matriz (110) para determinar uma matriz de autocorrelação r e um determinador do vetor do livro de códigos (120) para determinar o vetor do livro de códigos, dependendo da matriz de autocorrelação r. o determinador de matriz (110) é configurado para determinar a matriz de autocorrelação r pela determinação de coeficientes do vetor de um vetor r, caracterizado pela matriz de autocorrelação r compreender uma pluralidade de fileiras e uma pluralidade de colunas, em que o vetor r indica uma das colunas ou uma das fileiras da matriz de autocorrelação r, em que r(i, j) = r(ii- jl), em que r(i, j) indica os coeficientes da matriz de autocorrelação r, em que i é um primeiro índice que indica uma de uma pluralidade de fileiras da matriz de autocorrelação r e em que j é um segundo índice que indica uma da pluralidade de colunas da matriz de autocorrelação r. figura 1.

2. 发明专利

RU2707144C2 АУДИОКОДЕР И СПОСОБ ДЛЯ КОДИРОВАНИЯ АУДИОСИГНАЛА 未知
公开(公告)号：RU2707144C2
公开(公告)日：2019-11-22
申请号：RU2017135436
申请日：2016-04-06
申请人： FRAUNHOFER GES FORSCHUNG
发明人： BACKSTROM TOM , JOKINEN EMMA
IPC分类号： G10L19/08 , G10L19/032
摘要： Изобретениеотноситсяк средствамдлякодированияаудиосигнала. Техническийрезультатзаключаетсяв повышенииразборчивостидекодированногосигнала, когдаакустическийвходнойсигналискажаетсяфоновымшумоми другимиартефактами. Получаютинформациюо шуме, описывающуюшум, включенныйв аудиосигнал. Адаптивнокодируютаудиосигналв зависимостиотинформациио шуметакимобразом, чтоточностькодированиявышедлячастейаудиосигнала, которыеменеезатронутышумом, включеннымв аудиосигнал, чемдлячастейаудиосигнала, которыебольшезатронутышумом, включеннымв аудиосигнал, приэтомчастотныесоставляющие, которыеменееискаженышумом, квантуютсяс меньшейошибкой, тогдакаксоставляющие, которыевероятносодержатошибкиотшума, имеютболеенизкийвесв процессеквантования. 6 н. и 19 з.п. ф-лы, 7 ил.

3. 发明专利

CA2914771C APPARATUS AND METHOD FOR AUDIO SIGNAL ENVELOPE ENCODING, PROCESSING AND DECODING BY MODELLING A CUMULATIVE SUM REPRESENTATION EMPLOYING DISTRIBUTION QUANTIZATION AND CODING 未知
公开(公告)号：CA2914771C
公开(公告)日：2018-07-17
申请号：CA2914771
申请日：2014-06-10
申请人： FRAUNHOFER GES FORSCHUNG
发明人： BACKSTROM TOM , SCHUBERT BENJAMIN , MULTRUS MARKUS , DISCH SASCHA , SCHMIDT KONSTANTIN , PIETRZYK GRZEGORZ
IPC分类号： G10L19/06 , G10L19/03 , G10L19/032
摘要： An apparatus for generating an audio signal envelope from one or more coding values is provided. The apparatus comprises an input interface (1610) for receiving the one or more coding values, and an envelope generator (1620) for generating the audio signal envelope depending on the one or more coding values. The envelope generator (1620) is configured to generate an aggregation function depending on the one or more coding values, wherein the aggregation function comprises a plurality of aggregation points, wherein each of the aggregation points comprises an argument value and an aggregation value, wherein the aggregation function monotonically increases, and wherein each of the one or more coding values indicates at least one of an argument value and an aggregation value of one of the aggregation points of the aggregation function. Moreover, the envelope generator (1620) is configured to generate the audio signal envelope such that the audio signal envelope comprises a plurality of envelope points, wherein each of the envelope points comprises an argument value and an envelope value, and wherein an envelope point of the audio signal envelope is assigned to each of the aggregation points of the aggregation function such that the argument value of said envelope point is equal to the argument value of said aggregation point. Furthermore, the envelope generator (1620) is configured to generate the audio signal envelope such that the envelope value of each of the envelope points of the audio signal envelope depends on the aggregation value of at least one aggregation point of the aggregation function.

4. 发明专利

AU2011226143B2 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 未知
公开(公告)号：AU2011226143B2
公开(公告)日：2014-08-28
申请号：AU2011226143
申请日：2011-03-09
申请人： FRAUNHOFER GES FORSCHUNG , DOLBY INT AB
发明人： BAYER STEFAN , BACKSTROM TOM , GEIGER RALF , EDLER BERND , DISCH SASCHA , VILLEMOES LARS
IPC分类号： G10L21/04 , G10L19/02 , G10L19/022 , 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.

5. 发明专利

CA2942586C ENCODER, DECODER AND METHOD FOR ENCODING AND DECODING 未知
公开(公告)号：CA2942586C
公开(公告)日：2021-11-09
申请号：CA2942586
申请日：2015-03-03
申请人： FRAUNHOFER GES FORSCHUNG
发明人： BACKSTROM TOM , FISCHER JOHANNES , HELMRICH CHRISTIAN
IPC分类号： 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.

6. 发明专利

CA2914418A1 APPARATUS AND METHOD FOR AUDIO SIGNAL ENVELOPE ENCODING, PROCESSING AND DECODING BY SPLITTING THE AUDIO SIGNAL ENVELOPE EMPLOYING DISTRIBUTION QUANTIZATION AND CODING 未知
公开(公告)号：CA2914418A1
公开(公告)日：2014-12-18
申请号：CA2914418
申请日：2014-06-10
申请人： FRAUNHOFER GES FORSCHUNG
发明人： BACKSTROM TOM , SCHUBERT BENJAMIN , MULTRUS MARKUS , DISCH SASCHA , SCHMIDT KONSTANTIN , PIETRZYK GRZEGORZ
IPC分类号： G10L19/06 , G10L19/03 , G10L19/032
摘要： An apparatus for decoding to obtain a reconstructed audio signal envelope is provided. The apparatus comprises a signal envelope reconstructor (110) for generating the reconstructed audio signal envelope depending on one or more splitting points. Moreover, the apparatus comprises an output interface (120) for outputting the reconstructed audio signal envelope. The signal envelope reconstructor (110) is configured to generate the reconstructed audio signal envelope such that the one or more splitting points divide the reconstructed audio signal envelope into two or more audio signal envelope portions. An assignment rule defines a signal envelope portion value for each signal envelope portion of the two or more signal envelope portions depending on said signal envelope portion. Moreover, the signal envelope reconstructor (110) is configured to generate the reconstructed audio signal envelope such that, for each of the two or more signal envelope portions, an absolute value of its signal envelope portion value is greater than half of an absolute value of the signal envelope portion value of each of the other signal envelope portions.

7. 发明专利

MY168806A LINEAR PREDICTION BASED AUDIO CODING USING IMPROVED PROBABILITY DISTRIBUTION ESTIMATION 未知
公开(公告)号：MY168806A
公开(公告)日：2018-12-04
申请号：MYPI2014003598
申请日：2013-06-19
申请人： FRAUNHOFER GES FORSCHUNG
发明人： HELMRICH CHRISTIAN , FUCHS GUILLAUME , MULTRUS MARKUS , DIETZ MARTIN , BACKSTROM TOM
摘要： LINEAR PREDICTION BASED AUDIO CODING IS IMPROVED BY CODING A SPECTRUM COMPOSED OF A PLURALITY OF SPECTRAL COMPONENTS USING A PROBABILITY DISTRIBUTION ESTIMATION (28) DETERMINED FOR EACH OF THE PLURALITY OF SPECTRAL COMPONENTS FROM LINEAR PREDICTION COEFFICIENT INFORMATION. IN PARTICULAR, THE LINEAR PREDICTION COEFFICIENT INFORMATION IS AVAILABLE ANYWAY. ACCORDINGLY, IT MAY BE USED FOR DETERMINING THE PROBABILITY DISTRIBUTION ESTIMATION (28) AT BOTH ENCODING AND DECODING SIDE. THE LATTER DETERMINATION MAY BE IMPLEMENTED IN A COMPUTATIONALLY SIMPLE MANNER BY USING, FOR EXAMPLE, AN APPROPRIATE PARAMETERIZATION FOR THE PROBABILITY DISTRIBUTION ESTIMATION AT THE PLURALITY OF SPECTRAL COMPONENTS. ALL TOGETHER, THE CODING EFFICIENCY AS PROVIDED BY THE ENTROPY CODING IS COMPATIBLE WITH PROBABILITY DISTRIBUTION ESTIMATIONS AS ACHIEVED USING CONTEXT SELECTION, BUT ITS DERIVATION IS LESS COMPLEX. FOR EXAMPLE, THE DERIVATION MAY BE PURELY ANALYTICALLY AND/OR DOES NOT REQUIRE ANY INFORMATION ON ATTRIBUTES OF NEIGHBORING SPECTRAL LINES SUCH AS PREVIOUSLY CODED/DECODED SPECTRAL VALUES OF NEIGHBORING SPECTRAL LINES AS IS THE CASE IN SPATIAL CONTEXT SELECTION. FIGURE 1

8. 发明专利

BR112015007137A2 aparelho para codificar um sinal de fala que emprega acelp no domínio de autocorrelação 未知
公开(公告)号：BR112015007137A2
公开(公告)日：2017-07-04
申请号：BR112015007137
申请日：2013-07-31
申请人： FRAUNHOFER GES FORSCHUNG
发明人： BACKSTROM TOM , DIETZ MARTIN , FUCHS GUILLAUME , HELMRICH CHRISTIAN , MULTRUS MARKUS
IPC分类号： G10L19/107

9. 发明专利

AU2013283568B2 Linear prediction based audio coding using improved probability distribution estimation 未知
公开(公告)号：AU2013283568B2
公开(公告)日：2016-05-12
申请号：AU2013283568
申请日：2013-06-19
申请人： FRAUNHOFER GES FORSCHUNG
发明人： BACKSTROM TOM , HELMRICH CHRISTIAN , FUCHS GUILLAUME , MULTRUS MARKUS , DIETZ MARTIN
IPC分类号： G10L19/032 , G10L19/08
摘要： Linear prediction based audio coding is improved by coding a spectrum composed of a plurality of spectral components using a probability distribution estimation determined for each of the plurality of spectral components from linear prediction coefficient information. In particular, the linear prediction coefficient information is available anyway. Accordingly, it may be used for determining the probability distribution estimation at both encoding and decoding side. The latter determination may be implemented in a computationally simple manner by using, for example, an appropriate parameterization for the probability distribution estimation at the plurality of spectral components. All together, the coding efficiency as provided by the entropy coding is compatible with probability distribution estimations as achieved using context selection, but its derivation is less complex. For example, the derivation may be purely analytically and/or does not require any information on attributes of neighboring spectral lines such as previously coded/decoded spectral values of neighboring spectral lines as is the case in spatial context selection.

10. 发明专利

AU2014211524A1 Apparatus and method for synthesizing an audio signal, decoder, encoder, system and computer program 未知
公开(公告)号：AU2014211524A1
公开(公告)日：2015-09-17
申请号：AU2014211524
申请日：2014-01-28
申请人： FRAUNHOFER GES FORSCHUNG
发明人： FUCHS GUILLAUME , BACKSTROM TOM , GEIGER RALF , JAEGERS WOLFGANG , RAVELLI EMMANUEL
IPC分类号： G10L19/12 , G10L19/06 , G10L19/26
摘要： A method and an apparatus for synthesizing an audio signal are described. A spectral tilt is applied to the code of a codebook (202) used for synthesizing a current frame of the audio signal. The spectral tilt is based on the spectral tilt of the current frame of the audio signal. Further, an audio decoder operating in accordance with the inventive approach is described.

你已经成功收藏专利！

检索式保存成功!

IPRDB

热门服务

关于我们

友情链接

联系方式