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

Applicant: FRANCE TELECOM ,  KOVESI BALAZS ,  RAGOT STEPHANE ,  HOANG THI MINH NGUYET

Inventor： KOVESI BALAZS ,  RAGOT STEPHANE ,  HOANG THI MINH NGUYET

IPC: G10L19/00 ,  G10L19/008

CPC classification number: G10L19/008

Abstract: 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.

Abstract translation: 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。

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

Applicant: FRANCE TELECOM ,  KOVESI BALAZS ,  RAGOT STEPHANE

Inventor： KOVESI BALAZS ,  RAGOT STEPHANE

IPC: H04M1/253 ,  H03M13/05 ,  H04L12/18 ,  H04L29/12 ,  H04N7/15 ,  H04N7/26

CPC classification number: H03M13/451 ,  H03M13/09 ,  H03M13/356 ,  H03M13/6312

Abstract: 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.

Abstract translation: 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。

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

Applicant: FRANCE TELECOM ,  RAGOT STEPHANE ,  VIRETTE DAVID ,  KOVESI BALAZS

Inventor： RAGOT STEPHANE ,  VIRETTE DAVID ,  KOVESI BALAZS

IPC: G10L19/24 ,  G10L19/26

CPC classification number: G10L19/24 ,  G10L19/26

Abstract: 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.

Abstract translation: 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。

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

Applicant: FRANCE TELECOM ,  KOVESI BALAZS ,  RAGOT STEPHANE

Inventor： KOVESI BALAZS ,  RAGOT STEPHANE

IPC: G10L19/26 ,  G10L21/02

CPC classification number: G10L19/26 ,  G10L21/02

Abstract: 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).

Abstract translation: 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。

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

Applicant: FRANCE TELECOM ,  KOVESI BALAZS ,  RAGOT STEPHANE

Inventor： KOVESI BALAZS ,  RAGOT STEPHANE

IPC: G10L19/02 ,  G10L19/032

CPC classification number: G10L19/032 ,  G10L19/0212

Abstract: 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.

Abstract translation: 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。

公开(公告)号：WO2007107659A3

公开(公告)日：2008-12-18

申请号：PCT/FR2007050908

申请日：2007-03-09

Applicant: FRANCE TELECOM ,  RAGOT STEPHANE ,  GUILLAUME CYRIL

Inventor： RAGOT STEPHANE ,  GUILLAUME CYRIL

IPC: H04N7/26 ,  H03M7/30

CPC classification number: H03M7/3082 ,  H04N19/94

Abstract: The invention relates to a method for generating a dictionary for the vector quantisation of a signal. Said method comprises a step (10) of the statistical analysis of driving vectors representing the signal determining a finished set of code vectors (CITER) representing said driving vectors. The inventive method is characterised in that it also comprises a step (3) of modifying the finished set of code vectors in order to impose a minimum distance between the code vectors modified two-by-two, the set of said modified code vectors forming the dictionary.

Abstract translation: 本发明涉及一种用于生成信号矢量量化的字典的方法。 所述方法包括对表示确定表示所述驾驶向量的完成的代码矢量集合（CITER）的信号的驱动矢量的统计分析的步骤（10）。 本发明的方法的特征在于它还包括一个步骤（3），修改完成的代码矢量集合，以便在二乘二修改的代码矢量之间施加最小的距离，所述修改的代码矢量的集合形成 字典。

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

Applicant: FRANCE TELECOM ,  RAGOT STEPHANE ,  VIRETTE DAVID

Inventor： RAGOT STEPHANE ,  VIRETTE DAVID

IPC: G10L19/24

CPC classification number: G10L19/24

Abstract: 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.

Abstract translation: 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。

公开(公告)号：WO2010001020A3

公开(公告)日：2010-02-25

申请号：PCT/FR2009051064

申请日：2009-06-05

Applicant: FRANCE TELECOM ,  HOANG THI MINH NGUYET ,  RAGOT STEPHANE ,  OGER MARIE ,  ANTONINI MARC

Inventor： HOANG THI MINH NGUYET ,  RAGOT STEPHANE ,  OGER MARIE ,  ANTONINI MARC

IPC: H03M7/46 ,  G10L19/24

CPC classification number: H03M7/46 ,  G10L19/24 ,  H04N19/34 ,  H04N19/93

Abstract: The present invention relates to a technique of coding/decoding by bit planes, in which the whole components of a vector to be coded are decomposed into a binary representation in a succession of bit planes, from the plane of the most significant bits to the plane of the least significant bits. Within the meaning of the invention, the coding of the most significant bit plane is performed, while assigning a first type of coding symbol comprising at least two possible values of symbols ("+", "-") to represent a number of successive zeros in binary in a plane, and two other values (« 0 », « 1 ») for coding the sign of a significant bit, and the coding of the following bit planes up to the least significant bit plane is performed according to the steps of identifying the null bits in the planes already coded and extracting bits with the same positions in a current plane to be coded so as to form a nonsignificant part (Pk nonsig), of identifying the non-null bits in the planes already coded and extracting the bits with the same positions in the current plane to be coded so as to form a significant part (Pk sig), of coding the bits of the nonsignificant part using the first type of symbol, and of coding the bits of the significant part using a second type of symbol to code the value of the bits in the significant part.

Abstract translation: 本发明涉及一种通过位平面进行编码/解码的技术，其中待编码的矢量的整个分量在一系列比特平面中从最高有效位的平面分解成二进制表示， 的最低有效位。 在本发明的含义内，执行最高有效位平面的编码，同时分配包括符号（“+”，“ - ”）的至少两个可能值的第一类型的编码符号来表示连续零的数量 在平面中为二进制，以及用于对有效位的符号进行编码的另外两个值（“0”，“1”），并且根据以下步骤执行以下位平面到最低有效位平面的编码 识别已编码的平面中的零位，并提取要编码的当前平面中具有相同位置的位，以便形成非有效部分（Pk nonsig），以识别已编码的平面中的非空位，并提取 在当前平面中具有与要编码的位置相同的位，以便形成使用第一类型符号对无效部分的位进行编码的有效部分（Pk sig），以及使用 第二种类型的符号来代码 重要部分的位数值。

公开(公告)号：WO2007107670A2

公开(公告)日：2007-09-27

申请号：PCT/FR2007050959

申请日：2007-03-20

Applicant: FRANCE TELECOM ,  RAGOT STEPHANE ,  GUILLAUME CYRIL

Inventor： RAGOT STEPHANE ,  GUILLAUME CYRIL

IPC: G10L19/14 ,  G10L19/04 ,  G10L19/24 ,  G10L21/02

CPC classification number: G10L19/04 ,  G10L19/24 ,  G10L21/0364

Abstract: The invention relates to a method for post-processing, in an audio decoder, a signal reconstructed by the temporal and frequential shaping (805, 807) of an excitation signal obtained on the basis of at least one parameter in a first frequency band, said temporal and frequential shaping being carried out at least on the basis of a temporal envelope and a frequential envelope received and decoded (801, 802) in a second frequency band. The method is such that, once the shaping (805,807) has been carried out, steps of comparing the amplitude of the reconstructed signal with the received and decoded temporal envelope (s) are followed, and an amplitude compression is applied to the reconstructed signal if at least one threshold of the temporal envelope is exceeded. The invention relates to a post-processing module for implementing the inventive method, and to an audio decoder. It is used for transmitting and storing digital signals such as audiofrequency signals: speech, music, etc.

Abstract translation: 本发明涉及一种用于在音频解码器中对基于第一频带中的至少一个参数获得的激励信号的时间和频率整形（805,807）重构的信号进行后处理的方法，所述方法 至少基于在第二频带中接收和解码（801,802）的时间包络和频率包络来执行时间和频率整形。 该方法是，一旦完成了成形（805,807），则遵循将重构信号的幅度与接收和解码的时间包络进行比较的步骤，并将幅度压缩应用于重构信号，如果 超过了时间包络的至少一个阈值。 本发明涉及一种用于实现本发明的方法的后处理模块和一种音频解码器。 用于发送和存储诸如音频信号的数字信号：语音，音乐等。

公开(公告)号：WO2007093726A3

公开(公告)日：2007-10-18

申请号：PCT/FR2007050760

申请日：2007-02-07

Applicant: FRANCE TELECOM ,  RAGOT STEPHANE ,  TRILLING ROMAIN

Inventor： RAGOT STEPHANE ,  TRILLING ROMAIN

IPC: G10L19/02 ,  G10L19/24

CPC classification number: G10L19/24 ,  G10L19/0208

Abstract: The invention relates to a hierarchical audio encoder in a frequency band divided into a first sub-band and a second sub-band which are adjacent to each other, said encoder comprising: a core encoder (305) for encoding an original signal in the first sub-band of the frequency band; a calculation stage (306) for calculating a residual signal {e) from the original signal and from the signal supplied by the core encoder; and a device (307) for perceptual weighting of the residual signal {e). According to the invention, the perceptual weighting device comprises a perceptual weighting filter (307) with gain compensation that can perform the spectral continuity between the signal at the output of the perceptual weighting filter with gain compensation and the signal in the second sub-band. The invention can be applied to the transmission and storage of digital signals, such as the audio signals of speech, music, etc.

Abstract translation: 本发明涉及一种被划分为彼此相邻的第一子带和第二子带的频带中的分级音频编码器，所述编码器包括：核心编码器，用于对第一子带和第二子带中的原始信号进行编码 频段的子带; 用于从原始信号和由核心编码器提供的信号计算残差信号{e）的计算阶段（306） 以及用于对残余信号{e）进行感知加权的装置（307）。 根据本发明，感知加权装置包括具有增益补偿的感知加权滤波器（307），其可以利用增益补偿在感知加权滤波器的输出处的信号与第二子带中的信号之间执行频谱连续性。 本发明可以应用于数字信号的传输和存储，例如语音，音乐等的音频信号。