公开(公告)号：US20090292544A1

公开(公告)日：2009-11-26

申请号：US12309074

申请日：2007-06-19

Applicant: David Virette ,  Alexandre Guerin

Inventor： David Virette ,  Alexandre Guerin

IPC: G10L21/00

CPC classification number: H04S3/004 ,  G10L19/008 ,  H04S1/002 ,  H04S3/002 ,  H04S3/02 ,  H04S2400/01 ,  H04S2420/01 ,  H04S2420/03 ,  H04S2420/05

Abstract: The invention is aimed at improving the quality of the filtering by transfer functions of HRTF type of signals (L, R) compressed in a transformed domain, for binaural playing on two channels (L-BIN, R-BIN), using a combination of HRTF filters (hL,L, hL,R) including a decorrelated version (HRTF-C*, HRTF-E*) of a few of these filters. For this purpose, a decorrelation cue is given with spatialization parameters (SPAT) accompanying the compressed signals (L, R). The invention makes it possible to improve the broadening in the binaural rendition of audio scenes initially in a multi-channel format.

Abstract translation: 本发明旨在通过在变换域中压缩的信号（L，R）的HRTF类型的传递函数来提高滤波质量，用于双通道（L-BIN，R-BIN）上的双耳播放，使用 HRTF滤波器（hL，L，hL，R）包括几个滤波器的去相关版本（HRTF-C *，HRTF-E *）。 为此，给出了伴随压缩信号（L，R）的空间化参数（SPAT）的去相关提示。 本发明可以改善最初以多声道格式的音频场景的双耳再现的扩展。

公开(公告)号：US08880411B2

公开(公告)日：2014-11-04

申请号：US13120473

申请日：2009-10-05

Applicant: Pierrick Philippe ,  David Virette

Inventor： Pierrick Philippe ,  David Virette

IPC: G10L21/00 ,  G10L19/022 ,  H04R5/00 ,  H04B1/00 ,  G10L19/02 ,  G10L19/107 ,  G10L19/20 ,  G10L19/04

CPC classification number: G10L19/022 ,  G10L19/0212 ,  G10L19/04 ,  G10L19/107 ,  G10L19/20

Abstract: A method for encoding and decoding a digital audio signal is provided, said method comprising the steps of: encoding a first sequence of samples of the digital signal according to a transform encoding; encoding a second sequence of samples of the digital signal according to a predictive encoding; wherein the second sequence starts before the end of the first sequence, a subsequence common to the first and second sequences being thus encoded both by predictive encoding and by transform encoding.

Abstract translation: 提供了一种对数字音频信号进行编码和解码的方法，所述方法包括以下步骤：根据变换编码对数字信号的第一采样序列进行编码; 根据预测编码对数字信号的第二采样序列进行编码; 其中所述第二序列在所述第一序列结束之前开始，因此通过预测编码和通过变换编码对所述第一和第二序列共同的子序列进行编码。

公开(公告)号：US08417520B2

公开(公告)日：2013-04-09

申请号：US12446280

申请日：2007-10-17

Applicant: David Virette ,  Balazs Kovesi

Inventor： David Virette ,  Balazs Kovesi

IPC: G10L21/02 ,  G10L19/14

CPC classification number: G10L19/005 ,  G10L19/09

Abstract: 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. To this end, it proposes 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 optionally applying a correction of plus or minus a sample of the duration of this period (counted in terms of number of samples), by constituting 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 the effect of overvoicing in the synthesis of the generated signal is thereby attenuated.

Abstract translation: 本发明提出了由连续块组成的信号的合成。 更具体地，在接收到这样的信号时，通过合成替代该信号的丢失或错误的块。 为此，它提出了在信号合成生成期间的扩音器的衰减。 更具体地，基于在前一块估计或发送的音调周期（T），通过可选地应用该周期的持续时间的加法或减去的样本的校正来产生有声激励（以 样品），通过构成至少两个样品的组（A'，B'，C'，D'）和组中随机（B'，C'）或强制方式的样品的反转位置。 所产生的激励中的过度谐波因此被破坏，从而衰减了在产生的信号的合成中的超出结果的影响。

公开(公告)号：US08374853B2

公开(公告)日：2013-02-12

申请号：US11988758

申请日：2006-07-07

Applicant: Stéphane Ragot ,  David Virette

Inventor： Stéphane Ragot ,  David Virette

IPC: G10L19/00 ,  G10L19/12 ,  G10L19/14

CPC classification number: G10L19/24

Abstract: A system for coding a hierarchical audio signal, comprising, at least, a core layer using parametric coding by analysis by synthesis in a first frequency band, a band extension layer for widening said first frequency band into a second frequency band, or wideband. The system also comprises a wideband audio coding quality enhancement layer based on transform coding using a spectral parameter obtained from said band extension layer. Application to transmitting speech and/or audio signals over packet networks.

Abstract translation: 一种用于对分层音频信号进行编码的系统，至少包括使用第一频带中的合成分析的参数编码的核心层，用于将所述第一频带扩展成第二频带的带扩展层或宽带。 该系统还包括基于使用从所述频带扩展层获得的频谱参数的变换编码的宽带音频编码质量增强层。 应用于通过分组网络传输语音和/或音频信号。

公开(公告)号：US08253609B2

公开(公告)日：2012-08-28

申请号：US12809666

申请日：2008-12-11

Applicant: Pierrick Philippe ,  David Virette

Inventor： Pierrick Philippe ,  David Virette

IPC: H03M7/34

CPC classification number: H04N19/60 ,  G10L19/0212 ,  G10L19/022

Abstract: The present invention provides coding/decoding a digital signal, in particular using a transform with overlap employing weighting windows. In the invention, two consecutive and equal-size blocks of samples of the signal may be weighted by respective different successive windows. These two windows may be chosen independently of each other according to a criterion specific to the characteristics of the signal (entropy, data rate/distortion, etc.) that are determined for each of the two blocks.

Abstract translation: 本发明提供对数字信号的编码/解码，特别是使用加权窗口的重叠变换。 在本发明中，可以通过相应的不同连续窗口对信号的两个连续和相等大小的样本块进行加权。 这两个窗口可以根据针对两个块中的每一个确定的信号特性（熵，数据速率/失真等）的标准彼此独立地选择。

公开(公告)号：US20100324907A1

公开(公告)日：2010-12-23

申请号：US12446280

申请日：2007-10-17

Applicant: David Virette ,  Balazs Kovesi

Inventor： David Virette ,  Balazs Kovesi

IPC: G10L13/00

CPC classification number: G10L19/005 ,  G10L19/09

Abstract: 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. To this end, it proposes 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 optionally applying a correction of plus or minus a sample of the duration of this period (counted in terms of number of samples), by constituting 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 the effect of overvoicing in the synthesis of the generated signal is thereby attenuated.

Abstract translation: 本发明提出了由连续块组成的信号的合成。 更具体地，在接收到这样的信号时，通过合成替代该信号的丢失或错误的块。 为此，它提出了在信号合成生成期间的扩音器的衰减。 更具体地，基于在前一块估计或发送的音调周期（T），通过可选地应用该周期的持续时间的加法或减去的样本的校正来产生有声激励（以 样品），通过构成至少两个样品的组（A'，B'，C'，D'）和组中随机（B'，C'）或强制方式的样品的反转位置。 因此产生的激励中的过度谐波被破坏，从而衰减了在产生的信号的合成中的超出的效果。

公开(公告)号：US07792679B2

公开(公告)日：2010-09-07

申请号：US10582025

申请日：2004-11-24

Applicant: David Virette ,  Claude Lamblin ,  Abdellatif Benjelloun Touimi

Inventor： David Virette ,  Claude Lamblin ,  Abdellatif Benjelloun Touimi

IPC: G10L19/00

CPC classification number: G10L19/002 ,  G10L19/0212 ,  G10L19/12 ,  G10L19/18

Abstract: The invention relates to the compression coding of digital signals such as multimedia signals (audio or video), and more particularly a method for multiple coding, wherein several encoders each comprising a series of functional blocks receive an input signal in parallel. Accordingly, a method is provided in which, a) the functional blocks forming each encoder are identified, along with one or several functions carried out of each block, b) functions which are common to various encoders are itemized and c) said common functions are carried out definitively for a part of at least all of the encoders within at least one same calculation module.

Abstract translation: 本发明涉及诸如多媒体信号（音频或视频）之类的数字信号的压缩编码，更具体地说，涉及一种用于多重编码的方法，其中包括一系列功能块的几个编码器并行地接收输入信号。 因此，提供一种方法，其中a）识别形成每个编码器的功能块以及由每个块执行的一个或多个功能，b）各种编码器共同的功能被分项，c）所述共同功能是 在至少一个相同的计算模块内对至少所有的编码器的一部分进行明确地执行。

公开(公告)号：US20090204412A1

公开(公告)日：2009-08-13

申请号：US12224566

申请日：2007-02-13

Applicant: Balazs Kovesi ,  David Virette

Inventor： Balazs Kovesi ,  David Virette

IPC: G10L21/00

CPC classification number: G10L19/083 ,  G10L19/005

Abstract: Decoder for an audio signal coded by a coder including a long-term prediction filter wherein the decoder comprises: a block (211) for detecting transmission frame losses; a module (222) for calculating values of an error indication function representative of the cumulative adaptive excitation error during decoding following said transmission frame loss, an arbitrary value being assigned to said adaptive excitation gain for the lost frame; a module (213) for calculating an error indication parameter from said values of the error indication function; a comparator (214) for comparing said error indication parameter to at least one given threshold; and a discriminator (215) adapted to determine as a function of the results supplied by the comparator (214) a value of at least one adaptive excitation gain to be used by the decoder.

Abstract translation: 一种用于由包括长期预测滤波器的编码器编码的音频信号的解码器，其中解码器包括：用于检测传输帧损耗的块（211）; 模块（222），用于计算代表在所述传输帧丢失之后的解码期间的累积自适应激励误差的误差指示函数的值，任意值被分配给丢失帧的所述自适应激励增益; 模块（213），用于根据所述错误指示功能的值计算误差指示参数; 比较器（214），用于将所述误差指示参数与至少一个给定阈值进行比较; 以及鉴别器（215），其适于根据所述比较器（214）提供的至少一个自适应激励增益的值来确定所述解码器使用的至少一个自适应激励增益的值。

公开(公告)号：US20090083045A1

公开(公告)日：2009-03-26

申请号：US12293072

申请日：2007-03-08

Applicant: Manuel Briand ,  David Virette

Inventor： Manuel Briand ,  David Virette

IPC: G10L19/00

CPC classification number: G10L19/008 ,  G10L19/24

Abstract: A system and a method for the scalable coding of a multi-channel audio signal comprising a principal component analysis (PCA) transformation of at least two channels (L, R) of the audio signal into a principal component (CP) and at least one residual sub-component (r) by rotation defined by a transformation parameter (θ), comprising the following steps: formation of a frequency subband-based residual structure (Sfr) on the basis of the at least one residual sub-component (r), and definition of a coded audio signal (SC) comprising the principal component (CP), at least one residual structure (Sfr) of a frequency subband and the transformation parameter (θ).

Abstract translation: 一种用于多声道音频信号的可缩放编码的系统和方法，包括将音频信号的至少两个声道（L，R）的主成分分析（PCA）变换成主要分量（CP）和至少一个 通过由变换参数（θ）定义的旋转的残余子分量（r），包括以下步骤：基于所述至少一个残余子分量（r）形成基于频率子带的残差结构（Sfr） 以及包括主成分（CP）的编码音频信号（SC），频率子带的至少一个残留结构（Sfr）和变换参数（theta）的定义。

公开(公告)号：US20090083044A1

公开(公告)日：2009-03-26

申请号：US12293041

申请日：2007-03-08

Applicant: Manuel Briand ,  David Virette

Inventor： Manuel Briand ,  David Virette

IPC: G10L19/00

CPC classification number: G10L19/008

Abstract: A system and a method for coding by principal component analysis (PCA) of a multi-channel audio signal comprising the following steps: decomposing at least two channels (L, R) of said audio signal into a plurality of frequency sub-bands (1(b1), . . . , 1(bN), r(b1), . . . , r(bN)), calculating at least one transformation parameter (θ(b1), . . . , θ(bN)) as a function of at least some of said plurality of frequency sub-bands, transforming at least some of said plurality of frequency sub-bands into a plurality of frequency sub-components as a function of said at least one transformation parameter (θ(b1), . . . , θ(bN)), said plurality of frequency sub-components comprising principal frequency sub-components (CP(b1), . . . , CP(bN)), combining at least some of said principal frequency sub-components (CP(b1), . . . , CP(bN)) in order to form a principal component (CP), and defining a coded audio signal (SC) representing said multi-channel audio signal (C1, . . . ,CM), said coded audio signal (SC) comprising said principal component (CP) and said at least one transformation parameter (θ(b1), . . . , θ(bN)).

Abstract translation: 一种用于通过多声道音频信号的主成分分析（PCA）进行编码的系统和方法，包括以下步骤：将所述音频信号的至少两个声道（L，R）分解成多个频率子带（1 （b1），...，1（bN），r（b1），...，r（bN）），计算至少一个变换参数（theta（b1），...，theta（bN） 所述多个频率子带中的至少一些的功能是将所述多个频率子带中的至少一些变换为多个频率子分量作为所述至少一个变换参数（θ（b1））的函数， ，...，theta（bN）），所述多个频率子分量包括主频分量分量（CP（b1），...，CP（bN）），将至少一些所述主频分量 组件（CP（b1），...，CP（bN）），以形成主要组件（CP），并且定义表示所述多声道音频信号（C1，...， CM）说 包括所述主成分（CP）和所述至少一个变换参数（theta（b1））的编码音频信号（SC）。 。 。 ，theta（bN））。