公开(公告)号：US08880413B2

公开(公告)日：2014-11-04

申请号：US12309074

申请日：2007-06-19

Applicant: David Virette ,  Alexandre Guerin

Inventor： David Virette ,  Alexandre Guerin

IPC: G10L19/00 ,  H04R5/00 ,  H04S3/00 ,  H04S3/02 ,  G10L19/008 ,  H04S1/00

CPC classification number: H04S3/004 ,  G10L19/008 ,  H04S1/002 ,  H04S3/002 ,  H04S3/02 ,  H04S2400/01 ,  H04S2420/01 ,  H04S2420/03 ,  H04S2420/05

Abstract: The invention is aimed at improving the quality of the filtering by transfer functions of HRTF type of signals (L, R) compressed in a transformed domain, for binaural playing on two channels (L-BIN, R-BIN), using a combination of HRTF filters (hL,L, hL,R) including a decorrelated version (HRTF-C*, HRTF-E*) of a few of these filters. For this purpose, a decorrelation cue is given with spatialization parameters (SPAT) accompanying the compressed signals (L, R). The Decorrelation comprises applying a different phase shift to each subband of the input signal combined with addition of an overall delay. The invention makes it possible to improve the broadening in the binaural rendition of audio scenes initially in a multi-channel format.

Abstract translation: 本发明旨在通过在变换域中压缩的信号（L，R）的HRTF类型的传递函数来提高滤波质量，用于双通道（L-BIN，R-BIN）上的双耳播放，使用 HRTF滤波器（hL，L，hL，R）包括几个滤波器的去相关版本（HRTF-C *，HRTF-E *）。 为此，给出了伴随压缩信号（L，R）的空间化参数（SPAT）的去相关提示。 解相关包括对输入信号的每个子带应用不同的相移以及总延迟的加法。 本发明可以改善最初以多声道格式的音频场景的双耳再现的扩展。

公开(公告)号：US20090292544A1

公开(公告)日：2009-11-26

申请号：US12309074

申请日：2007-06-19

Applicant: David Virette ,  Alexandre Guerin

Inventor： David Virette ,  Alexandre Guerin

IPC: G10L21/00

CPC classification number: H04S3/004 ,  G10L19/008 ,  H04S1/002 ,  H04S3/002 ,  H04S3/02 ,  H04S2400/01 ,  H04S2420/01 ,  H04S2420/03 ,  H04S2420/05

Abstract: The invention is aimed at improving the quality of the filtering by transfer functions of HRTF type of signals (L, R) compressed in a transformed domain, for binaural playing on two channels (L-BIN, R-BIN), using a combination of HRTF filters (hL,L, hL,R) including a decorrelated version (HRTF-C*, HRTF-E*) of a few of these filters. For this purpose, a decorrelation cue is given with spatialization parameters (SPAT) accompanying the compressed signals (L, R). The invention makes it possible to improve the broadening in the binaural rendition of audio scenes initially in a multi-channel format.

Abstract translation: 本发明旨在通过在变换域中压缩的信号（L，R）的HRTF类型的传递函数来提高滤波质量，用于双通道（L-BIN，R-BIN）上的双耳播放，使用 HRTF滤波器（hL，L，hL，R）包括几个滤波器的去相关版本（HRTF-C *，HRTF-E *）。 为此，给出了伴随压缩信号（L，R）的空间化参数（SPAT）的去相关提示。 本发明可以改善最初以多声道格式的音频场景的双耳再现的扩展。

公开(公告)号：US09847085B2

公开(公告)日：2017-12-19

申请号：US13995718

申请日：2011-12-16

Applicant: Pierrick Philippe ,  David Virette

Inventor： Pierrick Philippe ,  David Virette

IPC: G10L19/00 ,  G10L19/26 ,  G10L19/02 ,  G10L19/022

CPC classification number: G10L19/00 ,  G10L19/02 ,  G10L19/0204 ,  G10L19/0212 ,  G10L19/022 ,  G10L19/26

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.

公开(公告)号：US08788555B2

公开(公告)日：2014-07-22

申请号：US13056154

申请日：2009-07-03

Applicant: Pierrick Philippe ,  David Virette

Inventor： Pierrick Philippe ,  David Virette

IPC: G06F17/17

CPC classification number: G10L19/0212 ,  H03H17/0264

Abstract: A method for updating the processing capacity of an encoder or decoder to use a modulated transform having a size greater than a predetermined initial size is provided, particularly, where the encoders or decoders are for storing an initial prototype filter defined by an ordered set of initial size coefficients. A step is provided for constructing a prototype filter of a size greater than the initial size to implement the modulated transform of the greater size by inserting at least one coefficient between two consecutive coefficients of the initial prototype filter.

Abstract translation: 提供了一种用于更新编码器或解码器以使用尺寸大于预定初始尺寸的调制变换的处理能力的方法，特别地，其中编码器或解码器用于存储由初始化的有序集合定义的初始原型滤波器 尺寸系数。 提供了一个步骤，用于构建大于初始尺寸的原型滤波器，以通过在初始原型滤波器的两个连续系数之间插入至少一个系数来实现较大尺寸的调制变换。

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：US20110129092A1

公开(公告)日：2011-06-02

申请号：US13056169

申请日：2009-07-03

Applicant: David Virette ,  Pierrick Philippe

Inventor： David Virette ,  Pierrick Philippe

IPC: H04R5/00

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：US20090326931A1

公开(公告)日：2009-12-31

申请号：US11988758

申请日：2006-07-07

Applicant: Stéphane Ragot ,  David Virette

Inventor： Stéphane Ragot ,  David Virette

IPC: G10L21/00 ,  G10L19/00

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：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：US08880411B2

公开(公告)日：2014-11-04

申请号：US13120473

申请日：2009-10-05

Applicant: Pierrick Philippe ,  David Virette

Inventor： Pierrick Philippe ,  David Virette

IPC: G10L21/00 ,  G10L19/022 ,  H04R5/00 ,  H04B1/00 ,  G10L19/02 ,  G10L19/107 ,  G10L19/20 ,  G10L19/04

CPC classification number: G10L19/022 ,  G10L19/0212 ,  G10L19/04 ,  G10L19/107 ,  G10L19/20

Abstract: A method for encoding and decoding a digital audio signal is provided, said method comprising the steps of: encoding a first sequence of samples of the digital signal according to a transform encoding; encoding a second sequence of samples of the digital signal according to a predictive encoding; wherein the second sequence starts before the end of the first sequence, a subsequence common to the first and second sequences being thus encoded both by predictive encoding and by transform encoding.

Abstract translation: 提供了一种对数字音频信号进行编码和解码的方法，所述方法包括以下步骤：根据变换编码对数字信号的第一采样序列进行编码; 根据预测编码对数字信号的第二采样序列进行编码; 其中所述第二序列在所述第一序列结束之前开始，因此通过预测编码和通过变换编码对所述第一和第二序列共同的子序列进行编码。

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。