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1. 发明申请

WO2007099244A2 METHOD FOR LIMITING ADAPTIVE EXCITATION GAIN IN AN AUDIO DECODER 审中-公开
标题翻译：在音频解码器中限制自适应激励增益的方法
公开(公告)号：WO2007099244A2
公开(公告)日：2007-09-07
申请号：PCT/FR2007050779
申请日：2007-02-13
申请人： FRANCE TELECOM , KOVESI BALAZS , VIRETTE DAVID
发明人： KOVESI BALAZS , VIRETTE DAVID
IPC分类号： G10L19/00 , G10L19/005 , G10L19/08 , G10L19/083
CPC分类号： G10L19/083 , G10L19/005
摘要： The invention concerns a decoder for an audio signal coded by an encoder comprising a long-term predictive filter. According to the invention, said decoder comprises: a block (211) for detecting losses of transmission frames, a module (222) for calculating values of an error indicating function, representing the accumulated error in decoding on the adaptive excitation following said transmission frame loss, an arbitrary value being assigned to said adaptive excitation for the lost frame, a module (213) for calculating an error indicating parameter based on said values of the error indicating function, a comparator (214) of said error indicating parameter with at least one given threshold, a discriminator (215) for determining based on of the result provided by the comparator (214) a value of at least one adaptive excitation gain to be used by the decoder. The invention is applicable to encoding and decoding digital signals such as audiofrequency signals.
摘要翻译：本发明涉及由编码器编码的音频信号的解码器，该编码器包括长期预测滤波器。根据本发明，所述解码器包括：用于检测传输帧丢失的块（211）;用于计算错误指示函数的值的模块（222），表示解码所述传输帧丢失之后的自适应激励的累积错误，为丢失帧分配给所述自适应激励的任意值，用于基于所述误差指示函数的值计算误差指示参数的模块（213），所述误差指示参数的比较器（214）具有至少一个给定阈值;鉴别器（215），用于基于由比较器（214）提供的结果来确定解码器要使用的至少一个自适应激励增益的值。本发明适用于对诸如音频信号的数字信号进行编码和解码。

2. 发明申请

WO2008047051A3 ATTENUATION OF OVERVOICING, IN PARTICULAR FOR GENERATING AN EXCITATION AT A DECODER, IN THE ABSENCE OF INFORMATION 审中-公开
标题翻译：特别是在解码器上产生激励的情况下，在没有信息的情况下使视频衰减
公开(公告)号：WO2008047051A3
公开(公告)日：2008-06-12
申请号：PCT/FR2007052188
申请日：2007-10-17
申请人： FRANCE TELECOM , VIRETTE DAVID , KOVESI BALAZS
发明人： VIRETTE DAVID , KOVESI BALAZS
IPC分类号： G10L19/00 , G10L11/04 , G10L19/005 , G10L19/09
CPC分类号： G10L19/005 , G10L19/09
摘要： The invention proposes the synthesis of a signal consisting of consecutive blocks. It proposes more particularly, on receipt of such a signal, to replace, by synthesis, lost or erroneous blocks of this signal. It proposes for this purpose an attenuation of the overvoicing during the generation of a signal synthesis. More particularly, a voiced excitation is generated on the basis of the pitch period (T) estimated or transmitted at the previous block, by possibly applying a correction of plus or minus a sample of the duration of this period (counted in terms of number of samples), by constructing groups (A',B',C',D') of at least two samples and inverting positions of samples in the groups, randomly (B',C') or in a forced manner. An over-harmonicity in the excitation generated is thus broken and, thereby, the effect of overvoicing in the synthesis of the signal generated is attenuated.
摘要翻译：本发明提出了由连续块组成的信号的合成。它更具体地提出，在收到这样的信号时，通过综合取代该信号丢失或错误的块。它为此提出了在产生信号合成过程中消除发票的衰减。更具体地说，根据在前一个模块估计或传输的基音周期（T），通过可能地应用正或负该周期持续时间样本的校正（以数量通过构造至少两个样本的组（A'，B'，C'，D'）并随机地（B'，C'）或以强制方式反转组中的样本的位置。因此所产生的激励中的过度调和因此被破坏，并且由此在生成的信号的合成中过度调度的效果被衰减。

3. 发明申请

WO2007010158A3 METHOD FOR SWITCHING RATE- AND BANDWIDTH-SCALABLE AUDIO DECODING RATE 审中-公开
标题翻译：用于切换速率和带宽可分级音频解码速率的方法
公开(公告)号：WO2007010158A3
公开(公告)日：2007-05-10
申请号：PCT/FR2006050697
申请日：2006-07-10
申请人： FRANCE TELECOM , RAGOT STEPHANE , VIRETTE DAVID , KOVESI BALAZS
发明人： RAGOT STEPHANE , VIRETTE DAVID , KOVESI BALAZS
IPC分类号： G10L19/24 , G10L19/26
CPC分类号： G10L19/24 , G10L19/26
摘要： The invention concerns a method for switching the decoding rate of an audio signal encoded by a multiple-rate audio coding system, said decoding including at least one step of post-processing dependent on the rate. The invention is characterized in that upon switching from an initial rate to a final rate, said method includes a step of transition by continuously shifting from a signal with initial rate to a signal of final rate, at least one of said signal being subjected to a post-processing. The invention is applicable to transmission of VOIP speech and/or audio signals on data packets.
摘要翻译：本发明涉及一种用于切换由多速率音频编码系统编码的音频信号的解码速率的方法，所述解码包括取决于速率的至少一个后处理步骤。本发明的特征在于，在从初始速率切换到最终速率时，所述方法包括通过从具有初始速率的信号连续地切换到最终速率信号来进行切换的步骤，所述信号中的至少一个经历后期处理。本发明适用于在数据分组上传输VOIP语音和/或音频信号。

4. 发明申请

WO2008081144A3 LOW-DELAY TRANSFORM CODING USING WEIGHTING WINDOWS 审中-公开
标题翻译：使用称重窗口的低延迟变换编码
公开(公告)号：WO2008081144A3
公开(公告)日：2008-09-18
申请号：PCT/FR2007052541
申请日：2007-12-18
申请人： FRANCE TELECOM , KOVESI BALAZS , VIRETTE DAVID , PHLIPPE PIERRICK
发明人： KOVESI BALAZS , VIRETTE DAVID , PHLIPPE PIERRICK
IPC分类号： H03M7/30 , G10L19/02 , G10L19/022
CPC分类号： G10L19/022
摘要： The invention relates to method for the transform coding/decoding of a digital audio signal represented by a succession of frames, using windows of different lengths. According to the invention, the coding method comprises the following steps, namely: trying to detect (51) a particular event, such as an attack, in a current frame (T i ); and, if said particular event is at least detected at the start of the current frame (53), directly applying a short window (54) in order to code (56) the current frame (T i ) without applying a transition window. Consequently, the coding method has a reduced delay in relation to the prior art. In addition, an ad hoc processing step is applied during decoding in order to compensate for the direct passage from a long window to a short window during coding.
摘要翻译：本发明涉及使用不同长度的窗口对由一连串帧表示的数字音频信号进行变换编码/解码的方法。根据本发明，编码方法包括以下步骤，即：尝试在当前帧（T i i i i）中检测（51）特定事件，例如攻击; 并且如果在当前帧（53）的起始处至少检测到所述特定事件，则直接施加短窗口（54）以便对（56）当前帧（T i）进行编码（56）而不应用转换窗口。因此，编码方法相对于现有技术具有减少的延迟。此外，在解码期间应用自组织处理步骤，以便在编码期间补偿从长窗口到短窗口的直接传递。

5. 发明申请

WO2011045549A8 OPTIMIZED PARAMETRIC STEREO DECODING 审中-公开
标题翻译：优化参数立体声解码
公开(公告)号：WO2011045549A8
公开(公告)日：2012-05-03
申请号：PCT/FR2010052193
申请日：2010-10-15
申请人： FRANCE TELECOM , KOVESI BALAZS , RAGOT STEPHANE , HOANG THI MINH NGUYET
发明人： KOVESI BALAZS , RAGOT STEPHANE , HOANG THI MINH NGUYET
IPC分类号： G10L19/00 , G10L19/008
CPC分类号： G10L19/008
摘要： The invention relates to a method of parametric decoding of a stereo digital audio signal, comprising a step of synthesizing (synth.) the stereo signal, per frequency sub-band, on the basis of a decoded mono signal of formula (I), arising from a downmix of the stereo signal and from spatial information parameters of the stereo signal, in such a way that the signals obtained have the following form: formula (II), wherein formula (III) and formula (IV) represent the channels of the synthesized signal, formula (V) and formula (VI) represent the signals dependent on the decoded mono signal, and c 1[ j ] and c 2[ j ] represent the gains. The gains are characterised in that they are calculated in the following way: formula (VII), wherein formula Î [ j ] is an amplitude ratio between the two channels of the stereo signal, arising from the decoded parameters. The invention also relates to a decoder implementing the method as described.
摘要翻译：本发明涉及一种对立体声数字音频信号进行参数解码的方法，该方法包括以下步骤：基于解码的公式（I）的单声道信号，合成（合成）每个频率子带的立体声信号，产生从立体声信号的下混合和立体声信号的空间信息参数，使得获得的信号具有以下形式：公式（II），其中式（III）和式（IV）表示合成信号，公式（V）和公式（VI）表示取决于解码单声道信号的信号，并且c 1 [j]和c 2 [j]表示增益。增益的特征在于它们以以下方式计算：公式（VII），其中公式Î[j]是由解码参数产生的立体声信号的两个声道之间的振幅比。本发明还涉及一种实现所述方法的解码器。

6. 发明申请

WO2009010672A2 LIMITATION OF DISTORTION INTRODUCED BY A POST-PROCESSING STEP DURING DIGITAL SIGNAL DECODING 审中-公开
标题翻译：数字信号解码过程中后处理引入的失真限制
公开(公告)号：WO2009010672A2
公开(公告)日：2009-01-22
申请号：PCT/FR2008051246
申请日：2008-07-04
申请人： FRANCE TELECOM , KOVESI BALAZS , RAGOT STEPHANE
发明人： KOVESI BALAZS , RAGOT STEPHANE
IPC分类号： G10L19/26 , G10L21/02
CPC分类号： G10L19/26 , G10L21/02
摘要： The invention relates to the processing of a digital signal originating from a decoder and a noise reduction post-processing step, including, in particular, limitation of distortion introduced by the post-processing step in order to deliver a corrected output signal (SOUT), assigning said corrected output signal (SOUT) with: a current amplitude having an intermediary value between a current amplitude value of the post-processed signal (SPOST) and a corresponding current amplitude value of the decoded signal (S'MIC), or the current amplitude of the post-processed signal (SPOST), according to the respective values of the current amplitude of the post-processed signal (SPOST) and by the corresponding current amplitude of the decoded signal (S'MIC).
摘要翻译：本发明涉及从解码器产生的数字信号的处理和降噪后处理步骤，特别地，包括通过后处理步骤引入的失真的限制，以便递送校正的输出信号（SOUT），通过以下方式分配所述校正输出信号（SOUT）：具有后处理信号（SPOST）的当前振幅值与解码信号（S'MIC）的相应电流振幅值之间的中间值的电流幅度或电流根据后处理信号（SPOST）的当前幅度的相应值和解码信号（S'MIC）的对应电流幅度，后处理信号（SPOST）的振幅。

7. 发明申请

WO2007096551A3 METHOD FOR BINARY CODING OF QUANTIZATION INDICES OF A SIGNAL ENVELOPE, METHOD FOR DECODING A SIGNAL ENVELOPE AND CORRESPONDING CODING AND DECODING MODULES 审中-公开
标题翻译：用于信号包络的量化指示的二进制编码的方法，用于解码信号包的方法和相应的编码和解码模块
公开(公告)号：WO2007096551A3
公开(公告)日：2007-11-01
申请号：PCT/FR2007050781
申请日：2007-02-13
申请人： FRANCE TELECOM , KOVESI BALAZS , RAGOT STEPHANE
发明人： KOVESI BALAZS , RAGOT STEPHANE
IPC分类号： G10L19/02 , G10L19/032
CPC分类号： G10L19/032 , G10L19/0212
摘要： The invention concerns a module (402) for binary coding of a signal envelope, comprising a coding module (502) of a first variable length mode. According to the invention, the coding module of a first mode incorporates an envelope saturation detector and said coding module (402) also comprises a second coding module (503) of a second mode, arranged parallel to the coding module (502) of the first mode, and a mode selector (504) capable of maintaining one of the two coding modes, based on a code length criterion and on the result derived from the envelope saturation detector. The invention is applicable to coding by audiofrequency signal transform.
摘要翻译：本发明涉及一种用于信号包络二进制编码的模块（402），包括第一可变长度模式的编码模块（502）。根据本发明，第一模式的编码模块包括包络饱和检测器，并且所述编码模块（402）还包括与第一模式（502）的编码模块（502）平行布置的第二模式的第二编码模块（503）模式和模式选择器（504），其能够基于码长度标准和从包络饱和检测器导出的结果来维持两种编码模式之一。本发明适用于通过音频信号变换进行编码。

8. 发明申请

WO2009080982A2 PROCESSING OF BINARY ERRORS IN A DIGITAL AUDIO BINARY FRAME 审中-公开
标题翻译：在数字音频二进制帧中处理二进制错误
公开(公告)号：WO2009080982A2
公开(公告)日：2009-07-02
申请号：PCT/FR2008052259
申请日：2008-12-10
申请人： FRANCE TELECOM , KOVESI BALAZS , RAGOT STEPHANE
发明人： KOVESI BALAZS , RAGOT STEPHANE
IPC分类号： H04M1/253 , H03M13/05 , H04L12/18 , H04L29/12 , H04N7/15 , H04N7/26
CPC分类号： H03M13/451 , H03M13/09 , H03M13/356 , H03M13/6312
摘要： The invention relates to a method of processing binary errors in a binary frame emanating from a digital audio coder, comprising a step of receiving a current binary frame liable to comprise binary errors. According to the invention, the binary frame comprises sensitive bits to be protected which are catalogued in at least one category according to the type of parameter that they code and the method furthermore comprises the steps of receiving protection bits, of reading the sensitive bits received in the current binary frame, the number of sensitive bits being lower than the number of bits of the binary frame, of detecting binary errors as a function of said protection bits received and of said sensitive bits received and in the event of detecting at least one erroneous bit in said binary frame, of modifying the current binary frame before decoding, as a function of the category in which the erroneous bit is catalogued. The invention also pertains to a device implementing the method according to the invention as well as to a decoder and a coding/decoding system comprising such a device.
摘要翻译：本发明涉及一种处理从数字音频编码器发出的二进制帧中的二进制错误的方法，包括接收当前包含二进制错误的二进制帧的步骤。根据本发明，二进制帧包括要被保护的敏感位，其根据它们编码的参数的类型在至少一个类别中编目，并且该方法还包括以下步骤：接收保护位，读取在检测作为接收的所述保护位的函数的二进制错误的当前二进制帧，敏感位的数量低于二进制帧的位数，并且在检测到至少一个错误的位置的情况下，检测二进制错误在所述二进制帧中，根据错误位被编目的类别的函数来修改解码之前的当前二进制帧。本发明还涉及实现根据本发明的方法的装置以及包括这种装置的解码器和编码/解码系统。

9. 发明申请

WO2007096552A3 METHOD FOR TRAINED DISCRIMINATION AND ATTENUATION OF ECHOES OF A DIGITAL SIGNAL IN A DECODER AND CORRESPONDING DEVICE 审中-公开
标题翻译：在解码器和相应装置中对数字信号的回波进行训练判断和衰减的方法
公开(公告)号：WO2007096552A3
公开(公告)日：2007-10-18
申请号：PCT/FR2007050786
申请日：2007-02-13
申请人： FRANCE TELECOM , KOVESI BALAZS , GUYADER ALAIN LE
发明人： KOVESI BALAZS , LE GUYADER ALAIN
IPC分类号： G10L19/24
CPC分类号： G10L19/24
摘要： The invention concerns a method for trained discrimination and attenuation of echoes of a digital audio signal generated from a transform coding, which consists, for each current frame of the signal, in: comparing (A) in real time, in at least one frequency band a variable derived from one characteristic of the echo generating signal with that of a non-echo generating signal at a threshold value, and deducing therefrom (B) the existence or non-existence (C) of an echo derived from the transform coding, discriminating the existence of the echo and defining (D) a false alarm zone in the high-energy parts of the digital audio signal, determining an initial processing and attenuating the echoes (E) in the parts complementary to the low-energy false alarm zone and inhibiting (F) the attenuation of echoes in the false alarm zone. The invention is applicable to the technology of coders/decoders in particular hierarchical coders/decoders.
摘要翻译：本发明涉及一种用于对从变换编码产生的数字音频信号的回声进行训练的区分和衰减的方法，该方法对于信号的每个当前帧包括：实时比较（A）至少一个频带从回波产生信号的一个特性与处于阈值的非回波产生信号的一个特性导出的变量，并由此推导（B）从变换编码导出的回波的存在或不存在（C），区分（D）数字音频信号的高能部分中的虚警区域，确定初始处理并衰减与低能量虚警区域互补的部分中的回声（E），以及抑制（F）虚警区中回声的衰减。本发明适用于特定分层编码器/解码器中的编码器/解码器的技术。

10. 发明申请

WO2007006958A2 METHOD AND DEVICE FOR ATTENUATING ECHOES OF A DIGITAL AUDIO SIGNAL DERIVED FROM A MULTILAYER ENCODER 审中-公开
标题翻译：用于衰减从多层编码器传送的数字音频信号的方法和装置
公开(公告)号：WO2007006958A2
公开(公告)日：2007-01-18
申请号：PCT/FR2006001659
申请日：2006-07-07
申请人： FRANCE TELECOM , KOVESI BALAZS , GUYADER ALAIN LE
发明人： KOVESI BALAZS , LE GUYADER ALAIN
IPC分类号： G10L19/14 , G10L19/02 , G10L19/025 , G10L19/24
CPC分类号： G10L19/025 , G10L19/24
摘要： The invention concerns a method for discriminating and attenuating echoes of a digital audio signal generated by a multilayer hierarchical encoding from a transform encoding, generating echoes, and from a predictive encoding, not generating echoes, consisting in at least one decoding process, for each current frame of the digital audio signal, in comparing (A), in real time, the value of a ratio of a representation of the temporal envelope of the signal derived from an echo generating encoding with that of the signal derived from a non echo generating encoding; and, if the value of said ratio is not less than a threshold value, concluding (B) therefrom the presence of an echo derived from the transform encoding in the current frame, otherwise, if the value of the ratio is less than said threshold value, concluding (C) therefrom the non existence of an echo derived from the transform encoding; attenuating (D), depending on the existence of echo, the signals derived from the echo generating transform encoding so as to reduce the audibility of the echoes. The invention is applicable to the use of devices attenuating echoes of a digital audio signal and of hierarchical encoders incorporating such devices.
摘要翻译：本发明涉及一种用于鉴别和衰减由多层分层编码产生的数字音频信号的回波的方法，该数字音频信号由变换编码产生回波，并从预测编码中产生回波，该预测编码包括至少一个解码过程，对于每个电流数字音频信号的帧，在实时比较（A）时，从回波产生编码导出的信号的时间包络的表示与从非回波产生编码导出的信号的比值的比值 ; 并且如果所述比率的值不小于阈值，则从其中得出（B）从当前帧中的变换编码导出的回波的存在，否则，如果该比值小于所述阈值从而得出（C）不存在从变换编码得到的回波; 取决于回波的存在，衰减（D）从回波产生变换编码得到的信号，以减少回声的可听性。本发明可应用于衰减数字音频信号的回波的设备的使用以及包含这种设备的分层编码器的使用。

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