公开(公告)号：US20120185255A1

公开(公告)日：2012-07-19

申请号：US13382786

申请日：2010-06-25

Applicant: David Virette ,  Stéphane Ragot ,  Balazs Kovesi ,  Pierre Berthet

Inventor： David Virette ,  Stéphane Ragot ,  Balazs Kovesi ,  Pierre Berthet

IPC: G10L21/00

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。

公开(公告)号：US20090306992A1

公开(公告)日：2009-12-10

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: G10L19/00

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：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：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：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。

公开(公告)号：US08812327B2

公开(公告)日：2014-08-19

申请号：US13382786

申请日：2010-06-25

Applicant: David Virette ,  Stéphane Ragot ,  Balazs Kovesi ,  Pierre Berthet

Inventor： David Virette ,  Stéphane Ragot ,  Balazs Kovesi ,  Pierre Berthet

IPC: G10L19/00 ,  G10L19/02 ,  G10L19/12 ,  G10L21/00 ,  G10L21/02 ,  G10L21/04 ,  G10L25/00 ,  G06F15/00 ,  H04N19/00

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。

公开(公告)号：US08180632B2

公开(公告)日：2012-05-15

申请号：US12224566

申请日：2007-02-13

Applicant: Balazs Kovesi ,  David Virette

Inventor： Balazs Kovesi ,  David Virette

IPC: G10L19/00 ,  G10L19/02

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：US20100076754A1

公开(公告)日：2010-03-25

申请号：US12448734

申请日：2007-12-18

Applicant: Balazs Kovesi ,  David Virette ,  Pierrick Philippe

Inventor： Balazs Kovesi ,  David Virette ,  Pierrick Philippe

IPC: G10L19/02

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。

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：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。