Coding/decoding of digital audio signals
    21.
    发明授权
    Coding/decoding of digital audio signals 有权
    数字音频信号的编码/解码

    公开(公告)号:US08812327B2

    公开(公告)日:2014-08-19

    申请号:US13382786

    申请日:2010-06-25

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

    Abstract: A method of hierarchical coding of a digital audio frequency input signal into several frequency sub-bands, including a core coding of the input signal according to a first throughput and at least one enhancement coding of higher throughput, of a residual signal. The core coding uses a binary allocation according to an energy criterion. The method includes for the enhancement coding: calculating a frequency-based masking threshold for at least part of the frequency bands processed by the enhancement coding; determining a perceptual importance per frequency sub-band as a function of the masking threshold and as a function of the number of bits allocated for the core coding; binary allocation of bits in the frequency sub-bands processed by the enhancement coding, as a function of the perceptual importance determined; and coding the residual signal according to the bit allocation. Also provided are a decoding method, a coder and a decoder.

    Abstract translation: 一种将数字音频输入信号分层编码成若干频率子带的方法,包括根据第一吞吐量的输入信号的核心编码和较高吞吐量的残余信号的至少一个增强编码。 核心编码根据能量标准使用二进制分配。 该方法包括用于增强编码:计算由增强编码处理的至少部分频带的基于频率的掩蔽阈值; 确定每个频率子带的感知重要性作为掩蔽阈值的函数,并且作为分配给核心编码的比特数的函数; 由增强编码处理的频率子带中的位的二进制分配作为确定的感知重要性的函数; 并根据比特分配对残差信号进行编码。 还提供了解码方法,编码器和解码器。

    FILTERING IN THE TRANSFORMED DOMAIN
    22.
    发明申请
    FILTERING IN THE TRANSFORMED DOMAIN 有权
    在变换域中进行过滤

    公开(公告)号:US20130282387A1

    公开(公告)日:2013-10-24

    申请号:US13995718

    申请日:2011-12-16

    Abstract: A method for processing a signal in the form of consecutive sample blocks, the method comprising filtering in a transformed domain of sub-bands, and particularly equalization processing, applied to a current block in the transformed domain, and filtering-adjustment processing that is applied in the transformed domain to at least one block adjacent to the current block.

    Abstract translation: 一种用于处理连续样本块形式的信号的方法,所述方法包括对变换域中的当前块应用的子带的变换域,特别是均衡处理进行滤波,以及应用滤波调整处理 在变换的域中至少一个与当前块相邻的块。

    Device and method for encoding by principal component analysis a multichannel audio signal
    23.
    发明授权
    Device and method for encoding by principal component analysis a multichannel audio signal 有权
    通过主成分分析来编码多通道音频信号的装置和方法

    公开(公告)号:US08370134B2

    公开(公告)日:2013-02-05

    申请号:US12293041

    申请日:2007-03-08

    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 (I(b1), . . . , I(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: 一种用于通过主要分量分析(PCA)编码多声道音频信号的系统和方法,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(I (b1),...,I(bN),r(b1),...,r(bN)),计算至少一个变换参数(&Thetas;(b1),...,&thetas;(bN) )作为所述多个频率子带中的至少一些的函数,将所述多个频率子带中的至少一些频率子带变换为多个频率子分量,作为所述至少一个变换参数的函数(“ (b1),...,...,...(bN)),所述多个频率子分量包括主频分量分量(CP(b1),...,CP(bN)), 主要频率子分量(CP(b1),...,CP(bN)),以形成主分量(CP),并且定义表示所述多声道音频信号(C1, ... ,CM),所述编码音频信号(SC)包括所述主分量(CP)和所述至少一个变换参数(“The”;(b1),...,&thetas;(bN))。

    Device and method for graduated encoding of a multichannel audio signal based on a principal component analysis
    24.
    发明授权
    Device and method for graduated encoding of a multichannel audio signal based on a principal component analysis 有权
    基于主成分分析的多通道音频信号的分级编码的装置和方法

    公开(公告)号:US08359194B2

    公开(公告)日:2013-01-22

    申请号:US12293072

    申请日:2007-03-08

    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)变换为主成分(CP)和至少一个 包括以下步骤:基于所述至少一个残余子分量(r)来形成基于频率子带的残余结构(Sfr),所述残余子分量(r)由转换参数(& )和包括主成分(CP)的编码音频信号(SC)的定义,频率子带的至少一个残留结构(Sfr)和变换参数(“thetas”)。

    Method for limiting adaptive excitation gain in an audio decoder
    25.
    发明授权
    Method for limiting adaptive excitation gain in an audio decoder 有权
    用于限制音频解码器中的自适应激励增益的方法

    公开(公告)号:US08180632B2

    公开(公告)日:2012-05-15

    申请号:US12224566

    申请日:2007-02-13

    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。

    Joint sound synthesis and spatialization
    26.
    发明授权
    Joint sound synthesis and spatialization 有权
    联合声音综合与空间化

    公开(公告)号:US08059824B2

    公开(公告)日:2011-11-15

    申请号:US12225097

    申请日:2007-03-01

    CPC classification number: G10H7/00 G10H2210/301 H04R2499/11 H04S3/002

    Abstract: The invention concerns a process for joint synthesis and spatialization of multiple sound sources in associated spatial positions, including: a) a step of assigning to each source at least one parameter (pi) representing an amplitude; b) a step of spatialization consisting in implementing an encoding into a plurality of channels, wherein each amplitude (pi) is duplicated to be multiplied to a specialization gain (gim), each spatialization gain being determined for one encoding channel (pgm) and for a source to be spatialized (Si); c) a step of grouping (R) the parameters multiplied by the gains (Pim), in respective channels (pg1, . . . , pgM), by applying a sum of said multiplied parameters (pim) on all the sources (Si) for each channel (pgm), and d) a step of parametric synthesis (SYNTH(I), . . . , SYNTH(M)) applied to each of the channels (pgm).

    Abstract translation: 本发明涉及一种用于在相关联的空间位置中联合合成和多个声源空间化的过程,包括:a)向每个源分配表示振幅的至少一个参数(pi)的步骤; b)空间化步骤,其包括将编码实现到多个通道中,其中每个振幅(pi)被复制以与特殊增益(gim)相乘,每个空间增益被确定用于一个编码通道(pgm),并且对于 空间化源(Si); c)通过在所有源(Si)上施加所述相乘参数(pim)的和来将(R)参数乘以增益(Pim)分组在各个通道(pg1,...,pgM)中的步骤, 对于每个通道(pgm),以及d)应用于每个通道(pgm)的参数合成(SYNTH(I),...,SYNTH(M))的步骤。

    ADVANCED ENCODING OF MULTI-CHANNEL DIGITAL AUDIO SIGNALS
    27.
    发明申请
    ADVANCED ENCODING OF MULTI-CHANNEL DIGITAL AUDIO SIGNALS 有权
    高级编码多通道数字音频信号

    公开(公告)号:US20110249822A1

    公开(公告)日:2011-10-13

    申请号:US13139611

    申请日:2009-12-11

    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。

    SPATIAL SYNTHESIS OF MULTICHANNEL AUDIO SIGNALS
    28.
    发明申请
    SPATIAL SYNTHESIS OF MULTICHANNEL AUDIO SIGNALS 有权
    多通道音频信号的空间综合

    公开(公告)号:US20110106543A1

    公开(公告)日:2011-05-05

    申请号:US12996406

    申请日:2009-06-16

    CPC classification number: G10L19/008 H04S3/008 H04S3/02 H04S2420/03

    Abstract: A method and associated device are provided for spatial synthesis of a sum signal to obtain at least two output signals, the sum signal as well as the spatialization parameters being output from a parametric coding by matrixing of an original multi-channel signal. The method comprises: decorrelation of the sum signal to obtain a decorrelated signal; applying a synthesis matrix, whose coefficients depend on the spatialization parameters, to the decorrelated signal and to the sum signal to obtain said output signals, wherein for at least one range of value of at least one spatialization parameter, the coefficients of the synthesis matrix are determined according to a criterion of minimizing a quantitative function, relating to the quantity of decorrelated signal in each of the output signals obtained by applying the synthesis matrix.

    Abstract translation: 提供了一种用于空间合成和信号以获得至少两个输出信号的方法和相关设备,该和信号以及通过原始多信道信号的矩阵化从参数编码输出的空间参数。 该方法包括:求和信号的去相关以获得解相关信号; 将其系数取决于空间化参数的合成矩阵应用于解相关信号和和信号以获得所述输出信号,其中对于至少一个空间参数的值的至少一个范围,合成矩阵的系数为 根据使定量函数最小化的标准,与通过应用合成矩阵获得的每个输出信号中去相关信号的量相关。

    LOW-DELAY TRANSFORM CODING USING WEIGHTING WINDOWS
    29.
    发明申请
    LOW-DELAY TRANSFORM CODING USING WEIGHTING WINDOWS 有权
    使用称重窗口的低延迟变换编码

    公开(公告)号:US20100076754A1

    公开(公告)日:2010-03-25

    申请号:US12448734

    申请日:2007-12-18

    CPC classification number: G10L19/022

    Abstract: The invention relates to transform coding/decoding of a digital audio signal represented by a succession of frames, using windows of different lengths. For the coding within the meaning of the invention, it is sought to detect (51) a particular event, such as an attack, in a current frame (Ti): and, at least if said particular event is detected at the start of the current frame (53), a short window (54) is directly applied in order to code (56) the current frame (Ti) without applying a transition window. Thus, the coding has a reduced delay in relation to the prior art. In addition, an ad hoc processing is applied during decoding in order to compensate for the direct passage from a long window to a short window during coding.

    Abstract translation: 本发明涉及使用不同长度的窗口的由一系列帧代表的数字音频信号的变换编码/解码。 对于本发明意义上的编码,寻求在当前帧(Ti)中检测(51)特定事件,例如攻击,并且至少如果在该帧的起始处检测到所述特定事件 当前帧(53),直接应用短窗口(54)以便在不应用转换窗口的情况下对当前帧(Ti)进行编码(56)。 因此,与现有技术相比,编码具有减小的延迟。 此外,在解码期间应用ad hoc处理,以补偿在编码期间从长窗口到短窗口的直接传递。

    Optimized multiple coding method
    30.
    发明申请
    Optimized multiple coding method 有权
    优化多重编码方法

    公开(公告)号:US20070150271A1

    公开(公告)日:2007-06-28

    申请号:US10582025

    申请日:2004-11-24

    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. According to the invention, a) the functional blocks (BF10, BFnN) 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. (BF1CC, BFnCC).

    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。 根据本发明,a)识别形成每个编码器的功能块(BF 10,BF nN)以及由每个块执行的一个或几个功能,b)各种编码器共同的功能被分项,c)所述共同 对至少一个相同的计算模块内的至少所有编码器的一部分进行明确的功能。 (BF 1 CC,BFnCC)。

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