Binaural spatialization of compression-encoded sound data utilizing phase shift and delay applied to each subband
    1.
    发明授权
    Binaural spatialization of compression-encoded sound data utilizing phase shift and delay applied to each subband 有权
    使用相移和延迟的压缩编码声音数据的双耳空间化应用于每个子带

    公开(公告)号:US08880413B2

    公开(公告)日:2014-11-04

    申请号:US12309074

    申请日:2007-06-19

    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。

    Binaural spatialization of compression-encoded sound data
    2.
    发明申请
    Binaural spatialization of compression-encoded sound data 有权
    压缩编码声音数据的双耳空间化

    公开(公告)号:US20090292544A1

    公开(公告)日:2009-11-26

    申请号:US12309074

    申请日:2007-06-19

    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。

    Method for binaural synthesis taking into account a room effect
    3.
    发明授权
    Method for binaural synthesis taking into account a room effect 有权
    考虑到房间效应的双耳合成方法

    公开(公告)号:US08045718B2

    公开(公告)日:2011-10-25

    申请号:US12225691

    申请日:2007-03-08

    CPC classification number: H04S1/005 H04S3/004 H04S2400/01

    Abstract: The invention concerns a method for three-dimensional spatialization of audio channels from a filter BRIR incorporating a theater effect. For a specific number N of samples corresponding to the size of the pulse response of the BRIR filter, it consists in breaking down (A) the BRIR filter into at least a set of delay and amplitude values associated with the times of arrival of reflections; extracting (B) on the number of B samples at least one spectral module of the BRIR filter; and constituting (C) from each successive delay, its amplitude and its spectral module associated with an elementary BRIR filter (BRIRe) directly applied to the audio channels in the time, frequency or transformed domain. The invention is applicable to binaural or multichannel spatialization.

    Abstract translation: 本发明涉及一种用于从包含剧院效果的滤波器BRIR进行音频通道三维空间化的方法。 对于与BRIR滤波器的脉冲响应的大小相对应的特定数量N的样本,其包括将BRIR滤波器分解成与反射到达时间相关联的至少一组延迟和振幅值; 对BIR滤波器的至少一个频谱模块提取(B)B个样本数; 并且从每个连续延迟构成(C),其幅度及其与基本BRIR滤波器(BRIRe)相关联的频谱模块直接应用于时间,频率或变换域中的音频信道。 本发明适用于双耳或多通道空间化。

    Method for Binaural Synthesis Taking Into Account a Room Effect
    4.
    发明申请
    Method for Binaural Synthesis Taking Into Account a Room Effect 有权
    双耳合成方法考虑到房间效应

    公开(公告)号:US20090103738A1

    公开(公告)日:2009-04-23

    申请号:US12225691

    申请日:2007-03-08

    CPC classification number: H04S1/005 H04S3/004 H04S2400/01

    Abstract: The invention concerns a method for three-dimensional spatialization of audio channels from a filter BRIR filter incorporating a theater effect. For a specific number N of samples corresponding to the size of the pulse response of the BRIR filter, it consists in breaking down (A) the BRIR filter into at least a set of delay and amplitude values associated with the times of arrival of reflections; extracting (B) on the number of B samples at least one spectral module of the BRIR filter; and constituting (C) from each successive delay, its amplitude and its spectral module associated with an elementary BRIR filter (BRIRe) directly applied to the audio channels in the time, frequency or transformed domain. The invention is applicable to binaural or multichannel spatialization.

    Abstract translation: 本发明涉及一种从包含剧院效果的滤波器BRIR滤波器对音频信道进行三维空间化的方法。 对于与BRIR滤波器的脉冲响应的大小相对应的特定数量N的样本,其包括将BRIR滤波器分解成与反射到达时间相关联的至少一组延迟和振幅值; 对BIR滤波器的至少一个频谱模块提取(B)B个样本数; 并且从每个连续延迟构成(C),其幅度及其与基本BRIR滤波器(BRIRe)相关联的频谱模块直接应用于时间,频率或变换域中的音频信道。 本发明适用于双耳或多通道空间化。

    Control of echo cancellation filters
    5.
    发明申请
    Control of echo cancellation filters 有权
    回波消除滤波器的控制

    公开(公告)号:US20080159552A1

    公开(公告)日:2008-07-03

    申请号:US12004489

    申请日:2007-12-21

    CPC classification number: H04M9/082

    Abstract: Coefficients of an adaptive filter representative of an acoustic channel between an emitted acoustic signal and a microphone signal are determined and smoothed in time. An echo is then estimated by filtering the emitted acoustic signal with the smoothed coefficients. Properties of the estimated echo and of the microphone signal are estimated. The echo cancellation filter is controlled as a function of a comparison between the properties of the estimated echo and those of the microphone signal so as to take into account the potential presence of a signal other than an echo signal in the microphone signal.

    Abstract translation: 表示发射的声信号和麦克风信号之间的声通道的自适应滤波器的系数在时间上被确定和平滑。 然后通过用平滑系数对发射的声信号进行滤波来估计回波。 估计回波和麦克风信号的性质。 回波消除滤波器作为估计回波的特性与麦克风信号的特性之间的比较来控制,以便考虑麦克风信号中除回波信号之外的信号的潜在存在。

    METHOD AND DEVICE FOR FILTERING DURING A CHANGE IN AN ARMA FILTER
    7.
    发明申请
    METHOD AND DEVICE FOR FILTERING DURING A CHANGE IN AN ARMA FILTER 有权
    在ARMA过滤器中更换过滤器的方法和装置

    公开(公告)号:US20140019504A1

    公开(公告)日:2014-01-16

    申请号:US14005801

    申请日:2012-03-14

    Abstract: A method and device are provided for filtering digital audio signals using at least one ARMA filter, particularly during a filter change. The method includes the following steps: a step of receiving a first request to change filtering to or from filtering by a first ARMA filter; and, in response to the first request, a step of gradually switching, at each of a plurality of cascaded first filtering blocks, between digital-signal filtering by a first basic filtering cell and digital-signal filtering by another associated basic filtering cell, the first basic filtering cells of the plurality of first filtering blocks factorizing the first filter.

    Abstract translation: 提供了一种用于使用至少一个ARMA滤波器来滤波数字音频信号的方法和装置,特别是在滤波器改变期间。 该方法包括以下步骤:接收向第一ARMA滤波器滤波或从第一ARMA滤波器滤波的第一请求的步骤; 并且响应于所述第一请求,在多个级联的第一滤波块中的每一个在由第一基本滤波单元进行数字信号滤波之间和由另一相关联的基本滤波单元进行数字信号滤波之间逐渐切换的步骤, 所述多个第一滤波块中的第一基本滤波单元对所述第一滤波器进行分解。

    Control of echo cancellation filters
    8.
    发明授权
    Control of echo cancellation filters 有权
    回波消除滤波器的控制

    公开(公告)号:US08150027B2

    公开(公告)日:2012-04-03

    申请号:US12004489

    申请日:2007-12-21

    CPC classification number: H04M9/082

    Abstract: Coefficients of an adaptive filter representative of an acoustic channel between an emitted acoustic signal and a microphone signal are determined and smoothed in time. An echo is then estimated by filtering the emitted acoustic signal with the smoothed coefficients. Properties of the estimated echo and of the microphone signal are estimated. The echo cancellation filter is controlled as a function of a comparison between the properties of the estimated echo and those of the microphone signal so as to take into account the potential presence of a signal other than an echo signal in the microphone signal.

    Abstract translation: 表示发射的声信号和麦克风信号之间的声通道的自适应滤波器的系数在时间上被确定和平滑。 然后通过用平滑系数对发射的声信号进行滤波来估计回波。 估计回波和麦克风信号的性质。 回波消除滤波器作为估计回波的特性与麦克风信号的特性之间的比较来控制,以便考虑麦克风信号中除回波信号之外的信号的潜在存在。

    Method and device for detecting acoustic shocks
    9.
    发明授权
    Method and device for detecting acoustic shocks 有权
    用于检测声学冲击的方法和装置

    公开(公告)号:US09031245B2

    公开(公告)日:2015-05-12

    申请号:US13807384

    申请日:2011-06-17

    CPC classification number: H04R29/00 G10L15/142 G10L25/00

    Abstract: A method and device are provided for detecting acoustic shocks in an audio stream. The method includes: breaking down the audio stream into audio frames; analyzing the audio frames in order to assign each audio frame a category value from among a plurality of predefined values; and determining the probability of an acoustic shock occurring in a current frame, based on a sequence of a given length of category values assigned to a set of frames, using a two-state Markov model, defined by a predetermined transition matrix and transmission matrix.

    Abstract translation: 提供了一种用于检测音频流中的声学冲击的方法和装置。 该方法包括:将音频流分解成音频帧; 分析音频帧以便从多个预定义值中分配每个音频帧的类别值; 并且使用由预定的转移矩阵和传输矩阵定义的两状态马尔科夫模型,基于分配给一组帧的给定类别值的长度的序列来确定在当前帧中发生声学冲击的概率。

    METHOD AND DEVICE FOR DETECTING ACOUSTIC SHOCKS
    10.
    发明申请
    METHOD AND DEVICE FOR DETECTING ACOUSTIC SHOCKS 有权
    用于检测声震的方法和装置

    公开(公告)号:US20130101124A1

    公开(公告)日:2013-04-25

    申请号:US13807384

    申请日:2011-06-17

    CPC classification number: H04R29/00 G10L15/142 G10L25/00

    Abstract: A method and device are provided for detecting acoustic shocks in an audio stream. The method includes: breaking down the audio stream into audio frames; analyzing the audio frames in order to assign each audio frame a category value from among a plurality of predefined values; and determining the probability of an acoustic shock occurring in a current frame, based on a sequence of a given length of category values assigned to a set of frames, using a two-state Markov model, defined by a predetermined transition matrix and transmission matrix.

    Abstract translation: 提供了一种用于检测音频流中的声学冲击的方法和装置。 该方法包括:将音频流分解成音频帧; 分析音频帧以便从多个预定义值中分配每个音频帧的类别值; 并且使用由预定的转移矩阵和传输矩阵定义的两状态马尔科夫模型,基于分配给一组帧的给定类别值的长度的序列来确定在当前帧中发生声学冲击的概率。

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