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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

WO2007110519A2 METHOD AND DEVICE FOR EFFICIENT BINAURAL SOUND SPATIALIZATION IN THE TRANSFORMED DOMAIN 审中-公开
标题翻译：变换域中有效双向声音空间化的方法和装置
公开(公告)号：WO2007110519A2
公开(公告)日：2007-10-04
申请号：PCT/FR2007050894
申请日：2007-03-08
申请人： FRANCE TELECOM , EMERIT MARC , PHILIPPE PIERRICK , VIRETTE DAVID
发明人： EMERIT MARC , PHILIPPE PIERRICK , VIRETTE DAVID
IPC分类号： H04S3/02
CPC分类号： H04S3/008 , H04S1/007
摘要： The invention concerns a method and a system for sound spatialization of a first set of not less than one of the audio channels encoded on of a number of frequency subbands (SBk) and decoded in a transformed domain (Fl, C, Fr, Sr, SI, Ife) into a second set of not less than two (Bl, Br) sound channels in the time domain, from modelling filters converted into a gain and a delay applicable in the transformed domain involving: filtering (A) through equalization, subband delay of the signal by applying at least one gain and one delay to generate from each of said encoded channels an equalized and delayed component; adding (B) a subset of equalized and delayed signals to create a number of filtered signals corresponding to not less than two; synthesizing (C) each of said filtered signals to obtain the second set of not less than two reproduction sound channels (Bl, Br) in the time domain.
摘要翻译：本发明涉及一种用于声音空间化的方法和系统，所述声音空间化用于在多个频率子带（SBk）上编码并且在变换域（F1，C，Fr，Sr， SI，Ife）转换成时域中不少于两个（B1，Br）声道的第二集合，从建模滤波器转换为适用于变换域的增益和延迟，包括：滤波（A）通过均衡，子带通过施加至少一个增益和一个延迟以从每个所述编码通道产生均衡和延迟的分量来延迟信号; 将（B）均衡和延迟信号的子集相加以产生对应于不少于两个的多个滤波信号; 合成（C）每个所述滤波信号以获得时域中不少于两个再现声道（B1，Br）的第二组。

6. 发明申请

WO2007007001A3 HIERARCHICAL ENCODING/DECODING DEVICE 审中-公开
标题翻译：分层编码/解码设备
公开(公告)号：WO2007007001A3
公开(公告)日：2007-04-12
申请号：PCT/FR2006050690
申请日：2006-07-07
申请人： FRANCE TELECOM , RAGOT STEPHANE , VIRETTE DAVID
发明人： RAGOT STEPHANE , VIRETTE DAVID
IPC分类号： G10L19/24
CPC分类号： G10L19/24
摘要： The invention concerns a hierarchical encoding system for an audio signal, comprising, at least one core parametric encoding core layer by analysis by synthesis in a first frequency band, a band extending layer designed to enlarge said first frequency band into a second frequency band, called extended band. The invention is characterized in that the system further comprises a layer for enhancing the audio encoding quality in the extended band, based on a transform encoding using a spectral parameter derived from said band extending layer. The invention is applicable to the transmission of speech and/or audio signals on packet networks.
摘要翻译：本发明涉及一种用于音频信号的分级编码系统，包括：通过在第一频带中的合成进行分析的至少一个核心参数编码核心层，被设计成将所述第一频带扩大为第二频带的频带扩展层，称为扩展频段本发明的特征在于，所述系统还包括用于基于使用从所述带延伸层导出的频谱参数的变换编码来增强扩展频带中的音频编码质量的层。本发明可应用于分组网络上的语音和/或音频信号的传输。

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