公开(公告)号：US09620139B2

公开(公告)日：2017-04-11

申请号：US13807657

申请日：2011-06-17

Applicant: Pierrick Philippe ,  David Virette ,  Claude Lamblin

Inventor： Pierrick Philippe ,  David Virette ,  Claude Lamblin

IPC: G10L21/00 ,  G10L19/06 ,  G10L13/00 ,  G06F11/00 ,  H03M7/30 ,  G10L19/18

CPC classification number: G10L21/00 ,  G10L19/06 ,  G10L19/18

Abstract: A method of coding/decoding of a digital audio signal comprising a succession of consecutive blocks of data, on the basis of a predictive filter. A modified predictive filter is used for the coding of at least one current block, the modified filter being constructed by the combination of: a rear filter calculated for a past block, preceding the current block, and enrichment parameters for the rear filter, which are determined as a function of the signal in the current block and comprising the coefficients of a modifying filter.

公开(公告)号：US08965775B2

公开(公告)日：2015-02-24

申请号：US13382794

申请日：2010-06-25

Applicant: David Virette ,  Pierre Berthet

Inventor： David Virette ,  Pierre Berthet

IPC: G10L19/00 ,  G10L19/24 ,  G10L19/002 ,  G10L19/02 ,  G10L19/038

CPC classification number: G10L19/24 ,  G10L19/002 ,  G10L19/0212 ,  G10L19/038

Abstract: A method of binary allocation in an enhancement coding/decoding for improving a hierarchical coding/decoding of digital audio signals, including a core coding/decoding in a first frequency band and a band extension coding/decoding in a second frequency band. For a predetermined number of bits to be allocated for the enhancement coding/decoding, a first number of bits is allocated to a coding/decoding for correcting the core coding/decoding in the first frequency band and according to a first mode of coding/decoding and a second number of bits is allocated to an enhancement coding/decoding for improving the extension coding/decoding in the second frequency band and according to a second mode of coding/decoding. Also provided are an allocation module implementing the method and a coder and decoder including this module.

Abstract translation: 一种用于改进数字音频信号的分层编码/解码的增强编码/解码中的二进制分配方法，包括第一频带中的核心编码/解码和第二频带中的频带扩展编码/解码。 对于要分配用于增强编码/解码的预定数量的比特，将第一数量的比特分配给用于校正第一频带中的核心编码/解码的编码/解码，并且根据编码/解码的第一模式 并且第二数量的比特被分配给用于改进第二频带中的扩展编码/解码的增强编码/解码，并且根据第二编码/解码模式。 还提供了实现该方法的分配模块和包括该模块的编码器和解码器。

公开(公告)号：US08630864B2

公开(公告)日：2014-01-14

申请号：US11989313

申请日：2006-07-10

Applicant: Stéphane Ragot ,  David Virette ,  Balazs Kovesi

Inventor： Stéphane Ragot ,  David Virette ,  Balazs Kovesi

IPC: G10L15/00

CPC classification number: G10L19/24 ,  G10L19/26

Abstract: A method of bitrate switching on decoding an audio signal coded by a audio coding system, said decoding comprising a post-processing step depending on the bitrate. On switching from an initial bitrate to a final bitrate, said method includes a transition step of continuous change from a signal at the initial bitrate to a signal at the final bitrate, one or both of said signals being post-processed. Application to transmission of VoIP speech and/or audio signals in data packet networks.

Abstract translation: 一种对由音频编码系统编码的音频信号进行解码的比特率切换的方法，所述解码包括取决于比特率的后处理步骤。 在从初始比特率切换到最终比特率时，所述方法包括从初始比特率的信号到最终比特率的信号的连续变化的转变步骤，所述信号中的一个或两个被后处理。 应用于数据分组网络中VoIP语音和/或音频信号的传输。

公开(公告)号：US08605909B2

公开(公告)日：2013-12-10

申请号：US12225677

申请日：2007-03-08

Applicant: Marc Emerit ,  Pierrick Philippe ,  David Virette

Inventor： Marc Emerit ,  Pierrick Philippe ,  David Virette

IPC: H04R5/00

CPC classification number: H04S3/008 ,  H04S1/007

Abstract: 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.

Abstract translation: 本发明涉及一种用于声音空间化的方法和系统，所述方法和系统用于在多个频率子带（SBk）上编码的不少于一个音频信道的第一组声音空间化，并在变换域（F1，C，Fr，Sr， SI，Ife）转换成时域中不少于两个（B1，Br）声道的第二组，从被转换成增益的建模滤波器和适用于变换域的延迟，涉及：滤波（A）通过均衡，子带 通过施加至少一个增益和一个延迟来从每个所述编码信道生成均衡和延迟的分量来延迟信号; 添加（B）均衡和延迟信号的子集以产生对应于不小于2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。

公开(公告)号：US20130103408A1

公开(公告)日：2013-04-25

申请号：US13807657

申请日：2011-06-17

Applicant: Pierrick Philippe ,  David Virette ,  Claude Lamblin

Inventor： Pierrick Philippe ,  David Virette ,  Claude Lamblin

IPC: G10L21/00

CPC classification number: G10L21/00 ,  G10L19/06 ,  G10L19/18

Abstract: A method of coding/decoding of a digital audio signal comprising a succession of consecutive blocks of data, on the basis of a predictive filter. A modified predictive filter is used for the coding of at least one current block, the modified filter being constructed by the combination of: a rear filter calculated for a past block, preceding the current block, and enrichment parameters for the rear filter, which are determined as a function of the signal in the current block and comprising the coefficients of a modifying filter.

Abstract translation: 一种基于预测滤波器对包括一系列连续的数据块的数字音频信号进行编码/解码的方法。 经修改的预测滤波器用于至少一个当前块的编码，修改的滤波器由以下组合构成：对于当前块之前的过去块计算的后滤波器和后滤波器的浓缩参数 作为当前块中的信号的函数确定并且包括修正滤波器的系数。

公开(公告)号：US08391373B2

公开(公告)日：2013-03-05

申请号：US12920352

申请日：2009-03-20

Applicant: David Virette ,  Pierrick Philippe ,  Balazs Kovesi

Inventor： David Virette ,  Pierrick Philippe ,  Balazs Kovesi

IPC: H04B14/04

CPC classification number: G10L19/005 ,  G10L19/0212 ,  G10L19/24

Abstract: A method is provided for concealing a transmission error in a digital signal chopped into a plurality of successive frames associated with different time intervals in which, on reception, the signal may comprise erased frames and valid frames, the valid frames comprising information relating to the concealment of frame loss. The method is implemented during a hierarchical decoding using a core decoding and a transform-based decoding using windows introducing a time delay of less than a frame with respect to the core decoding. The method includes concealing a first set of missing samples for the erased frame, implemented in a first time interval; a step of concealing a second set of missing samples utilizing information of said valid frame and implemented in a second time interval; and a step of transition between the first and the second set of missing samples to obtain at least part of the missing frame.

Abstract translation: 提供了一种隐藏在切成与不同时间间隔相关联的多个连续帧中的数字信号中的传输错误的方法，其中在接收时，信号可以包括擦除的帧和有效帧，该有效帧包括与隐藏有关的信息 的帧丢失。 该方法在使用核心解码和使用引入相对于核心解码的小于帧的时间延迟的窗口的基于变换的解码的分层解码期间实现。 该方法包括在第一时间间隔内隐藏针对被擦除的帧的第一组缺失样本集; 利用所述有效帧的信息隐藏第二组丢失样本并在第二时间间隔内实现的步骤; 以及在第一组和第二组缺失样本之间转换以获得缺失帧的至少一部分的步骤。

公开(公告)号：US20110249821A1

公开(公告)日：2011-10-13

申请号：US13139577

申请日：2009-12-11

Applicant: Florent Jaillet ,  David Virette

Inventor： Florent Jaillet ,  David Virette

IPC: H04R5/00

CPC classification number: G10L19/008

Abstract: A method for coding a multi-channel audio signal representing a sound scene comprising a plurality of sound sources is provided. This method comprises decomposing the multi-channel signal into frequency bands and, per frequency band, obtaining directivity information per sound source of the sound scene, the information being representative of the spatial distribution of the sound source in the sound scene, of selecting a set of sound sources of the sound scene constituting principal sources, of matrixing the selected principal sources to obtain a sum signal with a reduced number of channels and, of coding the directivity information and of forming a binary stream comprising the coded directivity information, the binary stream being transmittable in parallel with the sum signal. A decoding method is also provided that is able to decode the sum signal and the directivity information to obtain a multi-channel signal, to an adapted coder and an adapted decoder.

Abstract translation: 提供了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，并且每频带获得每个声场的每个声源的指向性信息，所述信息代表声场中的声源的空间分布，选择一组 构成主要源的声音场景的声源，对所选择的主要源进行矩阵化以获得具有减少的通道数的和信号，以及编码方向性信息和形成包括编码的方向性信息的二进制流，二进制流 可与和信号并行传输。 还提供了一种能够将求和信号和方向性信息解码以获得多信道信号的解码方法，其适用于编码器和适配解码器。

公开(公告)号：US20100283639A1

公开(公告)日：2010-11-11

申请号：US12809666

申请日：2008-12-11

Applicant: Pierrick Philippe ,  David Virette

Inventor： Pierrick Philippe ,  David Virette

IPC: H03M5/02

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：US20090232317A1

公开(公告)日：2009-09-17

申请号：US12225677

申请日：2007-03-08

Applicant: Marc Emerit ,  Pierrick Philippe ,  David Virette

Inventor： Marc Emerit ,  Pierrick Philippe ,  David Virette

IPC: H04R5/00

CPC classification number: H04S3/008 ,  H04S1/007

Abstract: 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.

Abstract translation: 本发明涉及一种用于声音空间化的方法和系统，所述方法和系统用于在多个频率子带（SBk）上编码的不少于一个音频信道的第一组声音空间化，并在变换域（F1，C，Fr，Sr， SI，Ife）转换成时域中不少于两个（B1，Br）声道的第二组，从被转换成增益的建模滤波器和适用于变换域的延迟，涉及：滤波（A）通过均衡，子带 通过施加至少一个增益和一个延迟来从每个所述编码信道生成均衡和延迟的分量来延迟信号; 添加（B）均衡和延迟信号的子集以产生对应于不小于2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。

公开(公告)号：US20070162236A1

公开(公告)日：2007-07-12

申请号：US10587907

申请日：2004-01-30

Applicant: Claude Lamblin ,  David Virette ,  Balazs Kovesi ,  Dominique Massaloux

Inventor： Claude Lamblin ,  David Virette ,  Balazs Kovesi ,  Dominique Massaloux

IPC: G06F19/00 ,  G06F7/38

CPC classification number: H03M7/3082 ,  G10L19/097 ,  H03M7/3088

Abstract: The invention relates to compression coding and/or decoding of digital signals, in particular by vector variable-rate quantisation defining a variable resolution. For this purpose an impulsion dictionary comprises: for a given dimension, increasing resolution dictionaries imbricated into each other and, for a given dimension, a union of: a totality (D′i ) of code-vectors produced, by inserting elements taken in a final set (A) into smaller dimension code-vectors according to a final set of predetermined insertion rules (F1) and a second totality of code-vectors (Y′) which are not obtainable by insertion into the smaller dimension code vectors according to said set of the insertion rules.

Abstract translation: 本发明涉及数字信号的压缩编码和/或解码，特别是通过定义可变分辨率的矢量可变速率量化。 为此目的，冲动字典包括：对于给定的维度，增加的分辨率词典彼此融合，并且对于给定的维度，通过插入元素的整数（D'i N）来产生代码矢量 根据预定插入规则（F 1）的最终集合和不能通过插入到较小维度代码中获得的第二总代码矢量（Y'），将最终集合（A）取入最小集合（A） 根据所述插入规则集合的向量。