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））。

公开(公告)号：US20070160216A1

公开(公告)日：2007-07-12

申请号：US10582834

申请日：2003-12-15

Applicant: Rozenn Nicol ,  David Virette ,  Marc Emerit

Inventor： Rozenn Nicol ,  David Virette ,  Marc Emerit

IPC: H04R5/00 ,  G10H1/46

CPC classification number: G10H1/0091 ,  G10H2210/301 ,  G10H2250/111

Abstract: The invention relates to the synthesis and the joint spatialization of sounds emitted by virtual sources. According to the invention, a step (ETA) is provided that consists of determining parameters including at least one gain (gi) for defining, at the same time, a loudness characterizing the nature of the virtual source and the position of the source relative to a predetermined origin.

Abstract translation: 本发明涉及由虚拟源发出的声音的合成和联合空间化。 根据本发明，提供了一种步骤（ETA），其包括确定包括至少一个增益（gi）的参数，该增益用于同时定义表征虚拟源的性质的响度以及源的相对于 预定来源。

公开(公告)号：US08867752B2

公开(公告)日：2014-10-21

申请号：US13056169

申请日：2009-07-03

Applicant: David Virette ,  Pierrick Philippe

Inventor： David Virette ,  Pierrick Philippe

IPC: H04R5/00 ,  G10L19/00 ,  G10L21/00 ,  G10L19/008 ,  H04S3/02 ,  G10L19/005

CPC classification number: G10L19/008 ,  G10L19/005 ,  H04R2420/03 ,  H04S3/02

Abstract: A method for processing sound data is provided for the reconstruction of multi-channel audio data on the basis at least of data on a reduced number of channels and of spatialization data. A test is carried out to determine whether the spatialization data received are valid. If the test is positive, a spatialization value is predicted according to a per respective model of a plurality of models. A prediction model is chosen on the basis of the spatialization values thus predicted and on the basis of the spatialization data received, to permit, in case of subsequent reception of defective spatialization data, a prediction according to this chosen model of a spatialization value and to use this predicted spatialization value for the reconstruction of the multi-channel audio data.

Abstract translation: 提供用于处理声音数据的方法，用于至少基于减少数量的信道和空间数据的数据的重建多声道音频数据。 进行测试以确定接收到的空间化数据是否有效。 如果测试是正的，则根据多个模型的每个相应模型来预测空间化值。 基于如此预测的空间化值，并且基于所接收的空间化数据来选择预测模型，以便在随后接收不合格的空间数据的情况下，根据该空间化值的所选择的模型进行预测，并且 使用该预测的空间化值来重建多声道音频数据。

公开(公告)号：US08817991B2

公开(公告)日：2014-08-26

申请号：US13139611

申请日：2009-12-11

Applicant: Florent Jaillet ,  David Virette

Inventor： Florent Jaillet ,  David Virette

IPC: H04R5/00 ,  G10L19/008 ,  H04S3/00 ,  H04S3/02

CPC classification number: G10L19/008 ,  H04S3/008 ,  H04S3/02 ,  H04S2400/11 ,  H04S2420/03 ,  H04S2420/07

Abstract: A method is provided for coding a multi-channel audio signal representing a sound scene comprising a plurality of sound sources. The method comprises decomposing the multi-channel signal into frequency bands and the following performed per frequency band: obtaining data representative of the direction of the sound sources of the sound scene, selecting a set of sound sources constituting principal sources, adapting the data representative of the direction of the selected principal sources, as a function of restitution characteristics of the multi-channel signal, determining a matrix for mixing the principal sources as a function of the adapted data, matrixing the principal sources by the matrix determined so as to obtain a sum signal with a reduced number of channels and coding the data representative of the direction of the sound sources and forming a binary stream comprising the coded data, the binary stream being transmittable in parallel with the sum signal.

Abstract translation: 提供一种用于编码表示包括多个声源的声音场景的多声道音频信号的方法。 该方法包括将多声道信号分解为频带，并且每个频带执行以下操作：获得表示声场的声源方向的数据，选择构成主要源的一组声源，使代表 所选择的主要源的方向作为多通道信号的恢复特性的函数，确定用于根据适配数据混合主要源的矩阵，通过确定的矩阵将主源矩阵化以获得 和信号，并且对代表声源方向的数据进行编码，并形成包括编码数据的二进制流，该二进制流可与和信号并行发送。