公开(公告)号：US09215544B2

公开(公告)日：2015-12-15

申请号：US12224840

申请日：2007-03-01

Applicant: Julien Faure ,  Jérôme Daniel ,  Marc Emerit

Inventor： Julien Faure ,  Jérôme Daniel ,  Marc Emerit

IPC: H04R5/00 ,  H04S1/00 ,  H04S5/00

CPC classification number: H04S1/00 ,  H04S1/002 ,  H04S5/00 ,  H04S2420/01

Abstract: The invention concerns sound spatialization with multichannel encoding for binaural reproduction on two loudspeakers, the spatial encoding being defined by encoding functions associated with multiple encoding channels and the decoding by applying filters for binaural reproduction. The invention provides for an optimization as follows: a) obtaining a original set of acoustic transfer functions particular to an individual's morphology (HRIR;HRTF), b) selecting spatial encoding functions (g(θ,φ,n)) and/or decoding filters (F(t,n)), and c) through successive iterations, optimizing the filters associated with the selected encoding functions or the encoding functions associated with the selected filters, or jointly the selected filters and encoding functions, by minimizing an error (c(HRIR,HRIR*)) calculated based on a comparison between: the original set of transfer functions (HRIR), and a set of reconstructed transfer functions (HRIR*) from encoding functions and decoding filters, whether optimized and/or selected.

Abstract translation: 本发明涉及用于在两个扬声器上进行双耳再现的多声道编码的声音空间化，空间编码由与多个编码信道相关联的编码功能定义，并通过应用用于双耳再现的滤波器进行解码。 本发明提供如下优化：a）获得特定于个人形态（HRIR; HRTF）的原始声学传递函数集合，b）选择空间编码函数（g（＆Theta，＆phgr;，n））和/ 或解码滤波器（F（t，n）），以及c）通过连续迭代，优化与所选择的编码功能相关联的滤波器或与所选择的滤波器相关联的编码功能，或联合地选择滤波器和编码功能， 基于来自编码功能和解码滤波器的原始传递函数集合（HRIR）和一组重构传递函数（HRIR *）计算的误差（c（HRIR，HRIR *）），无论优化和/或 选择。

公开(公告)号：US08059824B2

公开(公告)日：2011-11-15

申请号：US12225097

申请日：2007-03-01

Applicant: Grégory Pallone ,  Marc Emerit ,  David Virette

Inventor： Grégory Pallone ,  Marc Emerit ,  David Virette

IPC: H04S1/00 ,  H04R5/00

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。

公开(公告)号：US20090097663A1

公开(公告)日：2009-04-16

申请号：US12225097

申请日：2007-03-01

Applicant: Gregory Pallone ,  Marc Emerit ,  David Virette

Inventor： Gregory Pallone ,  Marc Emerit ,  David Virette

IPC: H04S1/00

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。

公开(公告)号：US20050018040A1

公开(公告)日：2005-01-27

申请号：US10495292

申请日：2002-11-12

Applicant: Georges Buchner ,  Marc Emerit ,  Jean-Phillippe Thomas ,  Alain Leyreloup

Inventor： Georges Buchner ,  Marc Emerit ,  Jean-Phillippe Thomas ,  Alain Leyreloup

IPC: H04N7/14 ,  H04N7/15

CPC classification number: H04N7/15 ,  H04N7/144

Abstract: The invention concerns a modular audio-visual system to bring together a local scene and a remote scene. The invention is characterized in that the system comprises: several concatenated audio-visual modules (M1, M2, M3), each module including an image sensing device and a sound pickup device for the local scene and, a device for restoring the image and the sound of the remote scene in an image plane (I), the modules being connected to a transmission network; automatic control and monitoring means (P) between the various modules to ensure continuity of the quality of the image and the sound on the image plane, when the people being filmed and recorded pass in front of the various image and sound pickup devices of the scene concerned.

Abstract translation: 本发明涉及一种将局部场景和远程场景结合在一起的模块化视听系统。 本发明的特征在于，该系统包括：多个连接的视听模块（M1，M2，M3），每个模块包括用于本地场景的图像感测装置和声音拾取装置，以及用于恢复图像和 在图像平面（I）中的远程场景的声音，模块连接到传输网络; 各个模块之间的自动控制和监视装置（P），以确保图像平面上的图像和声音的连续性，当被拍摄和记录的人通过场景的各种图像和拾音装置前面时 关心。

公开(公告)号：US08976972B2

公开(公告)日：2015-03-10

申请号：US13500955

申请日：2010-10-08

Applicant: Marc Emerit ,  Rozenn Nicol ,  Grégory Pallone

Inventor： Marc Emerit ,  Rozenn Nicol ,  Grégory Pallone

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

CPC classification number: H04S1/002 ,  G10L19/008 ,  H04S2420/01

Abstract: Processing of sound data encoded in a sub-band domain, for dual-channel playback of binaural or Transaural® type is provided, in which a matrix filtering is applied so as to pass from a sound representation with N channels with N>0, to a dual-channel representation. This sound representation with N channels comprises considering N virtual loudspeakers surrounding the head of a listener, and, for each virtual loudspeaker of at least some of the loudspeakers: a first transfer function specific to an ipsilateral path from the loudspeaker to a first ear of the listener, facing the loudspeaker, and a second transfer function specific to a contralateral path from said loudspeaker to the second ear of the listener, masked from the loudspeaker by the listener's head. The matrix filtering comprises a multiplicative coefficient defined by the spectrum, in the sub-band domain, of the second transfer function deconvolved with the first transfer function.

Abstract translation: 提供了编码在子带域中的声音数据的处理，用于双声道或超音速类型的双声道回放，其中应用矩阵滤波以便从具有N> 0的N个声道的声音表示传递到具有N> 0的声音表示， 双通道代表。 这种具有N个信道的声音表示包括考虑围绕收听者的头部的N个虚拟扬声器，并且对于至少一些扬声器的每个虚拟扬声器：对从扬声器到第一耳朵的同侧路径特有的第一传递函数 听众，面对扬声器，以及特定于从所述扬声器到收听者的第二耳朵的对侧路径的第二传递功能，由收听者的头部从扬声器屏蔽。 矩阵滤波包括由第一传递函数去卷积的第二传递函数在子带域中由频谱定义的乘法系数。

公开(公告)号：US20120201389A1

公开(公告)日：2012-08-09

申请号：US13500955

申请日：2010-10-08

Applicant: Marc Emerit ,  Rozenn Nicol ,  Grégory Pallone

Inventor： Marc Emerit ,  Rozenn Nicol ,  Grégory Pallone

IPC: H04R5/00

CPC classification number: H04S1/002 ,  G10L19/008 ,  H04S2420/01

Abstract: Processing of sound data encoded in a sub-band domain, for dual-channel playback of binaural or transaural® type is provided, in which a matrix filtering is applied so as to pass from a sound representation with N channels with N>0, to a dual-channel representation. This sound representation with N channels comprises considering N virtual loudspeakers surrounding the head of a listener, and, for each virtual loudspeaker of at least some of the loudspeakers: a first transfer function specific to an ipsilateral path from the loudspeaker to a first ear of the listener, facing the loudspeaker, and a second transfer function specific to a contralateral path from said loudspeaker to the second ear of the listener, masked from the loudspeaker by the listener's head. The matrix filtering comprises a multiplicative coefficient defined by the spectrum, in the sub-band domain, of the second transfer function deconvolved with the first transfer function.

Abstract translation: 提供了编码在子带域中的声音数据的处理，用于双耳或超音阶类型的双声道回放，其中应用矩阵滤波以便从具有N> 0的N个声道的声音表示传递到具有N> 0的声音表示， 双通道代表。 这种具有N个信道的声音表示包括考虑围绕收听者的头部的N个虚拟扬声器，并且对于至少一些扬声器的每个虚拟扬声器：对从扬声器到第一耳朵的同侧路径特有的第一传递函数 听众，面对扬声器，以及特定于从所述扬声器到收听者的第二耳朵的对侧路径的第二传递功能，由收听者的头部从扬声器屏蔽。 矩阵滤波包括由第一传递函数去卷积的第二传递函数在子带域中由频谱定义的乘法系数。

公开(公告)号：US20090067636A1

公开(公告)日：2009-03-12

申请号：US12224840

申请日：2007-03-01

Applicant: Julien Faure ,  Jerome Daniel ,  Marc Emerit

Inventor： Julien Faure ,  Jerome Daniel ,  Marc Emerit

IPC: H04R5/00

CPC classification number: H04S1/00 ,  H04S1/002 ,  H04S5/00 ,  H04S2420/01

Abstract: The invention concerns sound spatialization with multichannel encoding for binaural reproduction on two loudspeakers, the spatial encoding being defined by encoding functions associated with multiple encoding channels and the decoding by applying filters for binaural reproduction. The invention provides for an optimization as follows: a) obtaining a original set of acoustic transfer functions particular to an individual's morphology (HRIR;HRTF), b) selecting spatial encoding functions (g(θ,φ,n)) and/or decoding filters (F(t,n)), and c) through successive iterations, optimizing the filters associated with the selected encoding functions or the encoding functions associated with the selected filters, or jointly the selected filters and encoding functions, by minimizing an error (c(HRIR,HRIR*)) calculated based on a comparison between: the original set of transfer functions (HRIR), and a set of reconstructed transfer functions (HRIR*) from encoding functions and decoding filters, whether optimized and/or selected.

Abstract translation: 本发明涉及用于在两个扬声器上进行双耳再现的多声道编码的声音空间化，空间编码由与多个编码信道相关联的编码功能定义，并且通过应用用于双耳再现的滤波器进行解码。 本发明提供如下优化：a）获得特定于个人形态（HRIR; HRTF）的原始声学传递函数集合，b）选择空间编码函数（g（θ，phi，n））和/或解码 滤波器（F（t，n））和c）通过连续迭代，通过最小化误差来优化与所选择的编码函数或与所选择的滤波器相关联的编码功能或联合选择的滤波器和编码功能的滤波器 基于来自编码功能和解码滤波器的重组传递函数（HRIR *）的集合，基于优化和/或选择的原始传递函数集合（HRIR）和一组重构传递函数（HRIR *）计算的c（HRIR，HRIR *））。

公开(公告)号：US20070286430A1

公开(公告)日：2007-12-13

申请号：US11755563

申请日：2007-05-30

Applicant: Jean-Philippe Thomas ,  Marc Emerit

Inventor： Jean-Philippe Thomas ,  Marc Emerit

IPC: H04R3/12 ,  H04R3/00

CPC classification number: H04R27/00

Abstract: A method of comparison between pieces of information characterizing reference values and pieces of information characterizing current values of sound-reproducing systems of a system of (n) microphones mi and (p) speakers hpj for the control of said sound-reproducing systems characterized in that: A: for each speaker hpj, at least one sound signal S is sent on the speaker hpj, for each microphone mi, a piece of information hpjmi is retrieved, this piece of information characterizing the sound-reproducing system comprising the speaker hpj and the microphone mi, B: a reference matrix Qr is saved, this reference matrix being constituted by all the pieces of reference information hpjmi obtained following the sending of the sound signal S, C: as soon as a comparison is to be made, the step A is run with a sound signal S′ to obtain current information on a matrix Q, D: the matrices Q and Qr are compared.

Abstract translation: 一种用于表征参考值的信息和表征（n）个麦克风系统的声音再现系统的当前值的信息的信息的方法，（p）扬声器hp＆lt; j＆gt; 用于所述声音再现系统的控制，其特征在于：A：对于每个扬声器，在扬声器hp上发送至少一个声音信号S， SUB>，对于每个麦克风m 1，检索一条信息，其中表示声音再现的这条信息 系统包括扬声器hp和麦克风m，B：参考矩阵Q SUB R i被保存，该参考矩阵由全部 在发送声音信号S，C之后获得的参考信息片段：一旦进行比较，则步骤A被运行 具有声音信号S'以获得矩阵上的当前信息 Q，D：比较矩阵Q和Q 。

公开(公告)号：US20070160216A1

公开(公告)日：2007-07-12

申请号：US10582834

申请日：2003-12-15

Applicant: Rozenn Nicol ,  David Virette ,  Marc Emerit

Inventor： Rozenn Nicol ,  David Virette ,  Marc Emerit

IPC: H04R5/00 ,  G10H1/46

CPC classification number: G10H1/0091 ,  G10H2210/301 ,  G10H2250/111

Abstract: The invention relates to the synthesis and the joint spatialization of sounds emitted by virtual sources. According to the invention, a step (ETA) is provided that consists of determining parameters including at least one gain (gi) for defining, at the same time, a loudness characterizing the nature of the virtual source and the position of the source relative to a predetermined origin.

Abstract translation: 本发明涉及由虚拟源发出的声音的合成和联合空间化。 根据本发明，提供了一种步骤（ETA），其包括确定包括至少一个增益（gi）的参数，该增益用于同时定义表征虚拟源的性质的响度以及源的相对于 预定来源。

公开(公告)号：US20060198542A1

公开(公告)日：2006-09-07

申请号：US10547311

申请日：2004-02-18

Applicant: Abdellatif Benjelloun Touimi ,  Marc Emerit ,  Jean-Marie Pernaux

Inventor： Abdellatif Benjelloun Touimi ,  Marc Emerit ,  Jean-Marie Pernaux

IPC: H04R5/02

CPC classification number: H04S3/02 ,  H04S3/008 ,  H04S2400/01 ,  H04S2420/01

Abstract: The invention relates to the treatment of sound data for spatialized restitution of acoustic signals. At least one first and one second series of weighting terms are obtained for each acoustic signal, said terms representing a direction of perception of said acoustic signal by a listener. The acoustic signals are then applied to at least two sets of filtering units, which are disposed in parallel, in order to provide at least one first and one second output signal (L,R), corresponding to a linear combination of signals provided by said filtering units, which are respectively weighted by the weighting terms of the first and second series. According to the invention, each acoustic signal to be treated is at least partially compression coded and is expressed in the form of a vector of sub-signals associated with respective frequency sub-bands. Matrix filtering applied to each vector is carried out by each filtering unit in the space of the frequential sub-bands.

Abstract translation: 本发明涉及用于声学信号的空间恢复的声音数据的处理。 对于每个声信号获得至少一个第一和第二个加权项序列，所述项表示收听者对所述声信号的感知方向。 然后将声学信号施加到平行设置的至少两组滤波单元，以便提供至少一个第一和第二个输出信号（L，R），对应于由所述的信号提供的信号的线性组合 滤波单元，分别由第一和第二系列的加权项加权。 根据本发明，待处理的每个声信号至少部分地被压缩编码，并且以与各个频率子带相关联的子信号的向量的形式表示。 对每个矢量应用的矩阵滤波是由频率子带空间中的每个滤波单元进行的。