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1. 发明申请

WO2010003618A3 PROVIDING A TIME WARP ACTIVATION SIGNAL AND ENCODING AN AUDIO SIGNAL THEREWITH 审中-公开
标题翻译：提供时间激活信号和编码音频信号
公开(公告)号：WO2010003618A3
公开(公告)日：2010-03-25
申请号：PCT/EP2009004874
申请日：2009-07-06
申请人： FRAUNHOFER GES FORSCHUNG , BAYER STEFAN , DISCH SASCHA , GEIGER RALF , FUCHS GUILLAUME , NEUENDORF MAX , SCHULLER GERALD , EDLER BERND
发明人： BAYER STEFAN , DISCH SASCHA , GEIGER RALF , FUCHS GUILLAUME , NEUENDORF MAX , SCHULLER GERALD , EDLER BERND
IPC分类号： G10L19/10 , G10L21/04
CPC分类号： G10L21/04 , G10L19/002 , G10L19/0212 , G10L19/022 , G10L19/028 , G10L19/032 , G10L19/265 , G10L21/043 , G10L25/90
摘要： An audio encoder comprises a window function controller (504), a windower (502), a time warper (506) with a final quality check functionality, a time/frequency converter (508), a TNS stage (510) or a quantizer encoder (512), the window function controller (504), the time warper (506), the TNS stage (510) or an additional noise filling analyzer (524) are controlled by signal analysis results obtained by a time warp analyzer (516) or a signal classifier (520). Furthermore, a decoder applies a noise filling operation using a manipulated noise filling estimate depending on a harmonic or speech characteristic of the audio signal.
摘要翻译：音频编码器包括窗函数控制器（504），加窗器（502），具有最终质量检查功能的时间整形器（506），时间/频率转换器（508），TNS级（510）或量化器编码器（512），窗口功能控制器（504），时间整经器（506），TNS平台（510）或附加噪声填充分析器（524）由时间扭曲分析器（516）获得的信号分析结果或信号分类器（520）。此外，解码器使用根据音频信号的谐波或语音特性的操纵噪声填充估计来应用噪声填充操作。

2. 发明申请

WO2009121499A8 AUDIO TRANSFORM CODING USING PITCH CORRECTION 审中-公开
标题翻译：使用倾斜校正进行音频变换编码
公开(公告)号：WO2009121499A8
公开(公告)日：2010-02-25
申请号：PCT/EP2009002118
申请日：2009-03-23
申请人： FRAUNHOFER GES FORSCHUNG , EDLER BERND , DISCH SASCHA , GEIGER RALF , BAYER STEFAN , KRAEMER ULRICH , FUCHS GUILLAUME , NEUENDORF MAX , MULTRUS MARKUS , SCHULLER GERALD , POPP HARALD
发明人： EDLER BERND , DISCH SASCHA , GEIGER RALF , BAYER STEFAN , KRAEMER ULRICH , FUCHS GUILLAUME , NEUENDORF MAX , MULTRUS MARKUS , SCHULLER GERALD , POPP HARALD
IPC分类号： G10L19/02
CPC分类号： G10L19/0212 , G10L19/022
摘要： A processed representation of an audio signal having a sequence of frames is generated by sampling the audio signal within a first and a second frame of the sequence of frames, the second frame following the first frame, the sampling using information on a pitch contour of the first and the second frame to derive a first sampled representation. The audio signal is sampled within the second and the third frame, the third frame following the second frame in the sequence of frames. The sampling uses the information on the pitch contour of the second frame and information on a pitch contour of the third frame to derive a second sampled representation. A first scaling window is derived for the first sampled representation and a second scaling window is derived for the second sampled representation, the scaling windows depending on the samplings applied to derive the first sampled representations or the second sampled representation.
摘要翻译：具有帧序列的音频信号的处理表示通过对帧序列的第一帧和第二帧中的音频信号进行采样，第一帧之后的第二帧，使用关于第一帧的音调轮廓的信息的采样第一和第二帧以得出第一采样表示。音频信号在第二帧和第三帧内采样，第三帧在帧序列中的第二帧之后。采样使用关于第二帧的音调轮廓的信息和关于第三帧的音调轮廓的信息来导出第二采样表示。对于第一采样表示导出第一缩放窗口，并且针对第二采样表示导出第二缩放窗口，缩放窗口取决于应用于导出第一采样表示或第二采样表示的采样。

3. 发明申请

WO2010040522A2 MULTI-RESOLUTION SWITCHED AUDIO ENCODING/DECODING SCHEME 审中-公开
标题翻译：多分辨率切换音频编码/解码方案
公开(公告)号：WO2010040522A2
公开(公告)日：2010-04-15
申请号：PCT/EP2009007205
申请日：2009-10-07
申请人： FRAUNHOFER GES FORSCHUNG , VOICEAGE CORP , NEUENDORF MAX , BAYER STEFAN , LECOMTE JEREMIE , FUCHS GUILLAUME , ROBILLIARD JUIIEN , RETTELBACH NIKOLAUS , NAGEL FREDERIK , GEIGER RAIF , MULTRUS MARKUS , GRILL BERNHARD , GOURNAY PHILIPPE , SALAMI REDWAN
发明人： NEUENDORF MAX , BAYER STEFAN , LECOMTE JEREMIE , FUCHS GUILLAUME , ROBILLIARD JUIIEN , RETTELBACH NIKOLAUS , NAGEL FREDERIK , GEIGER RAIF , MULTRUS MARKUS , GRILL BERNHARD , GOURNAY PHILIPPE , SALAMI REDWAN
IPC分类号： G10L19/14 , G10L19/022 , G10L19/24
CPC分类号： G10L19/24 , G10L19/022
摘要： An audio encoder for encoding an audio signal comprises a first coding branch (400), the first coding branch comprising a first converter (410) for converting a signal from a time domain into a frequency domain. Furthermore, the audio encoder comprises a second coding branch (500) comprising a second time/frequency converter (523). Additionally, a signal analyzer (300/525) for analyzing the audio signal is provided. The signal analyzer, on the hand, determines whether an audio portion is effective in the encoder output signal as a first encoded signal from the first encoding branch or as a second encoded signal from a second encoding branch. On the other hand, the signal analyzer determines a time/frequency resolution to be applied by the converters (410, 523) when generating the encoded signals. An output interface includes, in addition to the first encoded signal and the second encoded signal, a resolution information identifying the resolution used by the first time/frequency converter and used by the second time/frequency converter.
摘要翻译：用于编码音频信号的音频编码器包括第一编码分支（400），第一编码分支包括用于将来自时域的信号转换成频域的第一转换器（410）。此外，音频编码器包括包括第二时间/频率转换器（523）的第二编码分支（500）。另外，提供了用于分析音频信号的信号分析器（300/525）。手持信号分析仪确定音频部分是否在编码器输出信号中作为来自第一编码分支的第一编码信号或来自第二编码分支的第二编码信号有效。另一方面，当生成编码信号时，信号分析器确定要由转换器（410,523）应用的时间/频率分辨率。除了第一编码信号和第二编码信号之外，输出接口包括识别由第一时间/频率转换器使用并由第二时间/频率转换器使用的分辨率的分辨率信息。

4. 发明申请

WO2008000316A8 AUDIO ENCODER, AUDIO DECODER AND AUDIO PROCESSOR HAVING A DYNAMICALLY VARIABLE HARPING CHARACTERISTIC 审中-公开
标题翻译：音频编码器，音频解码器和具有动态变化特征的音频处理器
公开(公告)号：WO2008000316A8
公开(公告)日：2009-02-26
申请号：PCT/EP2007004401
申请日：2007-05-16
申请人： FRAUNHOFER GES FORSCHUNG , HERRE JUERGEN , GRILL BERNHARD , MULTRUS MARKUS , BAYER STEFAN , KRAEMER ULRICH , HIRSCHFELD JENS , WABNIK STEFAN , SCHULLER GERALD
发明人： HERRE JUERGEN , GRILL BERNHARD , MULTRUS MARKUS , BAYER STEFAN , KRAEMER ULRICH , HIRSCHFELD JENS , WABNIK STEFAN , SCHULLER GERALD
IPC分类号： G10L19/00 , G10L19/20
CPC分类号： G10L19/20 , G10L19/265
摘要： An audio encoder, an audio decoder or an audio processor includes a filter (12) for generating a filtered audio signal, the filter having a variable warping characteristic, the characteristic being controllable in response to a time-varying control signal (16), the control signal indicating a small or no warping characteristic or a comparatively high warping characteristic. Furthermore, a controller (18) is connected for providing the time-varying control signal, which depends on the audio signal. The filtered audio signal can be introduced to an encoding processor (22) having different encoding algorithms, one of which is a coding algorithm adapted to a specific signal pattern. Alternatively, the filter is a post-filter receiving a decoded audio signal.
摘要翻译：音频编码器，音频解码器或音频处理器包括用于产生经滤波的音频信号的滤波器（12），滤波器具有可变翘曲特性，该特性可响应于时变控制信号（16）而被控制，指示变形特性小或不翘曲特性的控制信号。此外，控制器（18）被连接用于提供取决于音频信号的时变控制信号。滤波后的音频信号可以被引入具有不同编码算法的编码处理器（22），其中之一是适用于特定信号模式的编码算法。或者，滤波器是接收经解码的音频信号的后置滤波器。

5. 发明申请

WO2010003563A8 AUDIO ENCODER AND DECODER FOR ENCODING AND DECODING AUDIO SAMPLES 审中-公开
标题翻译：用于编码和解码音频样本的音频编码器和解码器
公开(公告)号：WO2010003563A8
公开(公告)日：2011-04-21
申请号：PCT/EP2009004651
申请日：2009-06-26
申请人： FRAUNHOFER GES FORSCHUNG , VOICEAGE CORP , LECOMTE JEREMIE , GOURNAY PHILIPPE , BAYER STEFAN , MULTRUS MARKUS , BESSETTE BRUNO , GRILL BERNHARD
发明人： LECOMTE JEREMIE , GOURNAY PHILIPPE , BAYER STEFAN , MULTRUS MARKUS , BESSETTE BRUNO , GRILL BERNHARD
IPC分类号： G10L19/02 , G10L19/14
CPC分类号： G10L19/20 , G10L19/022
摘要： An audio encoder (100) for encoding audio samples, comprising a first time domain aliasing introducing encoder (110) for encoding audio samples in a first encoding domain, the first time domain aliasing introducing encoder (110) having a first framing rule, a start window and a stop window. The audio encoder (100) further comprises a second encoder (120) for encoding samples in a second encoding domain, the second encoder (120) having a different second framing rule. The audio encoder (100) further comprises a controller (130) switching from the first encoder (110) to the second encoder (120) in response to characteristic of the audio samples, and for modifying the second framing rule in response to switching from the first encoder (110) to the second encoder (120) or for modifying the start window or the stop window of the first encoder (110), wherein the second framing rule remains unmodified.
摘要翻译：一种用于编码音频样本的音频编码器（100），包括用于对第一编码域中的音频样本进行编码的第一时域混叠引入编码器（110），具有第一成帧规则的第一时域混叠引入编码器（110）窗口和停止窗口。音频编码器（100）还包括用于对第二编码域中的采样进行编码的第二编码器（120），第二编码器（120）具有不同的第二成帧规则。音频编码器（100）还包括响应于音频样本的特性从第一编码器（110）切换到第二编码器（120）的控制器（130），并且响应于从第二编码器第一编码器（110）到第二编码器（120）或用于修改第一编码器（110）的起始窗口或停止窗口，其中第二成帧规则保持未修改。

6. 发明专利

AU2022235537A1 Multisignal audio coding using signal whitening as preprocessing 未知
公开(公告)号：AU2022235537A1
公开(公告)日：2022-10-13
申请号：AU2022235537
申请日：2022-09-20
申请人： FRAUNHOFER GES FORSCHUNG
发明人： FOTOPOULOU ELENI , MULTRUS MARKUS , DICK SASCHA , MARKOVIC GORAN , MABEN PALLAVI , KORSE SRIKANTH , BAYER STEFAN , DISCH SASCHA , HERRE JÜRGEN
IPC分类号： G10L19/008 , G10L19/03
摘要： Abstract A multisignal encoder for encoding at least three audio signals, comprises: a signal preproces sor (100) for individually preprocessing each audio signal to obtain at least three preprocessed audio signals, wherein the preprocessing is performed so that a preprocessed audio signal is whitened with respect to the signal before preprocessing; an adaptive joint signal processor (200) for performing a processing of the at least three preprocessed audio signals to obtain at least three jointly processed signals or at least two jointly processed signals and an unpro cessed signal; a signal encoder (300) for encoding each signal to obtain one or more encoded signals; and an output interface (400) for transmitting or storing an encoded multisignal audio signal comprising the one or more encoded signals, side information relating to the prepro cessing and side information relating to the processing. Fig. 5a 100 180 200 joint processing signal - adaptive joint side information preprocessor o signal processor (whitening) J . (e.g. also IGF) bit distribution signal encoder information 520- 300- (e.g. rate loop processor) preprocessing side information output interface encoded 400i multi-signal audio signal Fig. 5a joint processing side information, 510 520 532 534 536 individually preprocessing pairwise energy bit ••• encoded side processing scaling distribution .•• signals information side information information information - TNS info e channel pair side info bits - FDNS info e full M/S or dual mono or - LTP info band-wise M/S and M/S mask - window info e IGF data Fig. 5b

7. 发明专利

ZA202103741B APPARATUS, METHOD AND COMPUTER PROGRAM FOR ENCODING, DECODING, SCENE PROCESSING AND OTHER PROCEDURES RELATED TO DIRAC BASED SPATIAL AUDIO CODING USING DIFFUSE COMPENSATION 未知
公开(公告)号：ZA202103741B
公开(公告)日：2022-06-29
申请号：ZA202103741
申请日：2021-05-31
申请人： FRAUNHOFER GES FORSCHUNG
发明人： FUCHS GUILLAUME , THIERGART OLIVER , KORSE SRIKANTH , DÖHLA STEFAN , MULTRUS MARKUS , KÜCH FABIAN , BOUTHÉON ALEXANDRE , EICHENSEER ANDREA , BAYER STEFAN
摘要： An apparatus for generating a sound field description using an input signal having a mono-signal or a multi-channel signal comprises: an input signal analyzer for analyzing the input signal to derive direction data and diffuseness data; a low-order components generator for generating a low-order sound field description from the input signal up to a predetermined order and mode; a mid-order components generator for generating a mid-order sound field description above the predetermined order or at the predetermined order and above the predetermined mode and below or at a first truncation order using a synthesis of at least one direct portion and of at least one diffuse portion using the direction data and the diffuseness data; and a high-order components generator for generating a high-order sound field description having a component above the first truncation order using a synthesis of at least one direct portion.

9. 发明专利

ES2907377T3 Aparato, procedimiento y programa informático para la codificación, la decodificación, el procesamiento de escenas y otros procedimientos relacionados con la codificación de audio espacial basada en DirAC 未知
公开(公告)号：ES2907377T3
公开(公告)日：2022-04-25
申请号：ES18779381
申请日：2018-10-01
申请人： FRAUNHOFER GES FORSCHUNG
发明人： FUCHS GUILLAUME , HERRE JÜRGEN , KÜCH FABIAN , DÖHLA STEFAN , MULTRUS MARKUS , THIERGART OLIVER , WÜBBOLT OLIVER , GHIDO FLORIN , BAYER STEFAN , JAEGERS WOLFGANG
IPC分类号： G10L19/008 , G10L19/16
摘要： Un aparato para la generación de una descripción de una escena de audio combinada, que comprende: una interfaz de entrada (100) para la recepción de una primera descripción de una primera escena en un primer formato y una segunda descripción de una segunda escena en un segundo formato, en el que el segundo formato es diferente del primer formato; un conversor de formatos (120) para la conversión de la primera descripción en un formato común y para la conversión de la segunda descripción en el formato común, cuando el segundo formato es diferente del formato común; y un combinador de formatos (140) para la combinación de la primera descripción en el formato común y la segunda descripción en el formato común para obtener la escena de audio combinada.

10. 发明专利

BR112021025265A2 Sintetizador de áudio, codificador de áudio, sistema, método e unidade de armazenamento não transitória 未知
公开(公告)号：BR112021025265A2
公开(公告)日：2022-03-15
申请号：BR112021025265
申请日：2020-06-15
申请人： FRAUNHOFER GES FORSCHUNG
发明人： BOUTHÉON ALEXANDRE , KÜCH FABIAN , FUCHS GUILLAUME , HERRE JÜRGEN , MULTRUS MARKUS , THIERGART OLIVER , DISCH SASCHA , BAYER STEFAN
IPC分类号： G10L19/008
摘要： sintetizador de áudio, codificador de áudio, sistema, método e unidade de armazenamento não transitória. são revelados vários exemplos de técnica de codificação e decodificação. em particular, um sintetizador de áudio (300) para gerar um sinal de síntese (336, 340, yr) a partir de um sinal de mixagem de redução (246, x) que compreende: uma interface de entrada (312) para receber o sinal de mixagem de redução (246, x), em que o sinal de mixagem de redução (246, x) tem diversos canais de mixagem de redução e informações externas (228), em que as informações externas (228) incluem informações de correlação e nível de canal (314, (csi), (qui)) de um sinal original (212, y), em que o sinal original (212, y) tem diversos canais originais; e um processador de síntese (404) para gerar, de acordo com pelo menos uma regra de mixagem, o sinal de síntese (336, 340, yr) que usa: informações de correlação e nível de canal (220, 314, (csi), (qui)) do sinal original (212, y); e informações de covariância (cx) associadas ao sinal de mixagem de redução (324, 246, x).

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