METHOD AND APPARATUS FOR ESTIMATING SPATIAL CONTENT OF SOUNDFIELD AT DESIRED LOCATION

    公开(公告)号:US20210227327A1

    公开(公告)日:2021-07-22

    申请号:US17164443

    申请日:2021-02-01

    Abstract: In general, the present invention relates to a method and apparatus for estimating spatial content of a soundfield at a desired location, including a location that has actual sound content obstructed or distorted. According to certain aspects, the present invention aims at presenting a more natural, spatially accurate sound, for example to a user at the desired location who is wearing a helmet, mimicking the sound a user would experience if they were not wearing any headgear. Modes for enhanced spatial hearing may be applied which would include situation-dependent processing for augmented hearing. According to other aspects, the present invention aims at remotely reproducing the soundfield at a desired location with faithful reproduction of the spatial content of the soundfield.

    Method and system for artificial reverberation employing dispersive delays
    2.
    发明授权
    Method and system for artificial reverberation employing dispersive delays 有权
    使用分散延迟的人造混响的方法和系统

    公开(公告)号:US08391504B1

    公开(公告)日:2013-03-05

    申请号:US12006448

    申请日:2007-12-31

    CPC classification number: G10H1/0091 G10H3/186 G10H2210/285 G10H2250/535

    Abstract: The present invention relates to audio signal processing, and more particularly to methods and apparatuses for emulating and controlling various features of mechanical spring reverberation in a digital audio processing system. According to certain aspects of the invention, such an emulation is performed so as to enhance or alter the characteristics of a digitally stored or processed audio signal in substantially the same manner as a mechanical spring reverberation system. In one example embodiment, the propagation of energy through a mechanical spring is simulated using dispersive waveguides, wherein left-going and right-going waves are separately processed, and the effects of dispersion and attenuation commuted to the waveguide ends. According to additional aspects, many spring reverberators contain spring elements arranged in parallel, with no coupling between springs. Accordingly, in another embodiment of the present invention, such reverberators are modeled using a set of waveguide structures, arranged in parallel, and tuned to simulate the dispersion and attenuation of the torsional propagation modes of each of the individual spring elements. According to further aspects, reverberators occasionally have spring elements comprised of spring segments connected in series. Accordingly, in yet another embodiment of the invention, such arrangements are emulated using dispersive waveguide structures with scattering junctions between modeled spring segments. According to still other embodiments of the invention, both longitudinal and torsional waves are simulated so as to produce a widening over time of successive arrivals at the simulated pick-up, to thereby account for the difference in propagation speed between the torsional and longitudinal modes.

    Abstract translation: 音频信号处理技术领域本发明涉及音频信号处理,更具体地涉及用于模拟和控制数字音频处理系统中的机械弹簧混响的各种特征的方法和装置。 根据本发明的某些方面,执行这种仿真以便以与机械弹簧混响系统基本相同的方式增强或改变数字存储或处理的音频信号的特性。 在一个示例实施例中,通过使用分散波导来模拟通过机械弹簧的能量的传播,其中分开处理左后波和右波,并且波导的扩散和衰减的影响结束。 根据另外的方面,许多弹簧混响器包含平行布置的弹簧元件,弹簧之间没有联接。 因此,在本发明的另一个实施例中,这样的混响器使用一组平行布置的波导结构进行建模,并被调谐以模拟各个弹簧元件的扭转传播模式的分散和衰减。 根据其他方面,混响器偶尔具有由串联连接的弹簧段组成的弹簧元件。 因此,在本发明的另一个实施例中,使用在建模的弹簧段之间具有散射接头的分散波导结构来仿真这种布置。 根据本发明的其它实施例,模拟纵向和扭转波,以便在模拟拾取器处产生连续到达的随着时间的加宽,从而考虑扭转模式和纵向模式之间的传播速度的差异。

    Digital resonant shelf filter
    3.
    发明授权
    Digital resonant shelf filter 失效
    数字谐振架滤波器

    公开(公告)号:US07831645B1

    公开(公告)日:2010-11-09

    申请号:US11249162

    申请日:2005-10-11

    CPC classification number: H03H17/0211

    Abstract: A method and system for designing a discrete-time filter having a transfer function which approximates that of an analog shelf filter is disclosed. Prior art methods include applying the bilinear transform to the analog filter, which has the drawback of warping high-frequency features of the desired transfer function. In an embodiment of the present invention, an analog filter is designed which anticipates the warping imposed by the bilinear transform. For filters whose features approach the Nyquist limit, the inventive method provides a closer approximation to the analog response than direct application of the bilinear transform.

    Abstract translation: 公开了一种用于设计离散时间滤波器的方法和系统,该离散时间滤波器具有近似于模拟机架滤波器的传递函数。 现有技术的方法包括将双线性变换应用于模拟滤波器,该模拟滤波器具有翘曲期望传输函数的高频特征的缺点。 在本发明的一个实施例中,设计了一种模拟滤波器,其预测双线性变换施加的翘曲。 对于其特征接近奈奎斯特极限的滤波器,本发明的方法比直接应用双线性变换提供了更接近于模拟响应的近似。

    Crosstalk canceler
    4.
    发明授权
    Crosstalk canceler 失效
    串扰消除器

    公开(公告)号:US07263193B2

    公开(公告)日:2007-08-28

    申请号:US10691211

    申请日:2003-10-21

    Inventor: Jonathan S. Abel

    CPC classification number: H04S1/002 H04S3/00 H04S2400/01

    Abstract: The invention is a crosstalk canceler wherein different frequency bands are canceled at different locations so as to allow greater listener movement about the “sweet spot” while maintaining effective crosstalk cancellation. A spectrally smooth canceler equalization is used, reducing artifacts for listeners away from the sweet spot and further enlarging the sweet spot. Finally, the canceler equalization is adapted to either the anticipated or the actual crosscoherence among the input channels, producing a natural equalization regardless of the input.

    Abstract translation: 本发明是一种串扰消除器,其中不同频带在不同位置被取消,以便在维持有效的串扰消除的同时允许更大的收听者关于“甜蜜点”的移动。 使用光谱平滑的消除器均衡,减少远离甜点的听众的伪影,并进一步扩大甜蜜点。 最后,消除器均衡适应输入通道之间的预期或实际交叉相干,产生自然均衡,而与输入无关。

    Method and apparatus for processing signals to extract narrow bandwidth
features
    5.
    发明授权
    Method and apparatus for processing signals to extract narrow bandwidth features 失效
    用于处理信号以提取窄带宽特征的方法和装置

    公开(公告)号:US5430690A

    公开(公告)日:1995-07-04

    申请号:US121332

    申请日:1993-09-13

    Inventor: Jonathan S. Abel

    CPC classification number: G01R23/16

    Abstract: A method and apparatus for processing a set of signals to identify narrow bandwidth features of the signals, and optionally to process the signals further to extract information about the identified narrow-band features. The invention processes a set of input signal frames (a multi-dimensional pixel array) to generate a narrow-band feature signal (also a multi-dimensional pixel array) from which narrow-band features of the input signal frames can be efficiently, automatically, and unambiguously identified. In a class of preferred embodiments, the input signal frames are the power spectra of a set of sequentially measured signals. Thus, the set of input signal frames is a "spectrogram," comprising rows and columns of pixels (with row indices representing time, and column indices representing frequency). Alternatively, the input signal frames represent a data array of another type, such as a correlogram or a sequence of images. In a class of embodiments, the input signal frames are processed to identify narrow-band pixels (the subset of input signal frame pixels which represent narrow-band energy, or narrow-band processes). Then, the identified narrow-band pixels (which can be displayed as a narrow-band pixel array) undergo "feature extraction" processing to generate the output narrow-band feature signal (which can be displayed as the output image).

    Abstract translation: 一种用于处理一组信号以识别信号的窄带宽特征的方法和装置,并且可选地进一步处理信号以提取关于所识别的窄带特征的信息。 本发明处理一组输入信号帧(多维像素阵列)以产生窄带特征信号(也是多维像素阵列),从而可以自动地高效地输入输入信号帧的窄带特征 ,并明确确定。 在一类优选实施例中,输入信号帧是一组顺序测量信号的功率谱。 因此,该组输入信号帧是“谱图”,包括行和列的像素(行索引表示时间,列索引表示频率)。 或者,输入信号帧表示另一类型的数据阵列,例如相关图或图像序列。 在一类实施例中,处理输入信号帧以识别窄带像素(表示窄带能量或窄带处理的输入信号帧像素的子集)。 然后,识别的窄带像素(可以显示为窄带像素阵列)经历“特征提取”处理,以产生输出窄带特征信号(其可以显示为输出图像)。

    Method and system for artificial reverberation using modal decomposition

    公开(公告)号:US11049482B1

    公开(公告)日:2021-06-29

    申请号:US16384266

    申请日:2019-04-15

    Inventor: Jonathan S. Abel

    Abstract: In general, the present invention relates to a method and system for synthesizing artificial reverberation using modal analysis of a room or resonating object. In one embodiment of the inventive system, a collection of resonant filters is employed, each driven by the source signal, and their outputs summed. With filter resonance frequencies and dampings tuned to the modal frequencies and decay times of the acoustic space or resonating object being simulated, and filter gains set according to the source and listener positions within the space or object, any number of acoustic spaces and resonant objects may be simulated.

    Artificial reverberator room size control

    公开(公告)号:US10559295B1

    公开(公告)日:2020-02-11

    申请号:US16213993

    申请日:2018-12-07

    Inventor: Jonathan S. Abel

    Abstract: In music recording and virtual reality applications, it is desired to control the perceived size of a synthesized acoustic space. In an embodiment, a room size parameter is used to modify characteristics of an artificial reverberator so as to affect a listener's sense of the size of the acoustic space. Properties of the reverberation related to perceived room size such as decay time and equalization are adjusted interactively.

    Method and system for artificial reverberation using modal decomposition

    公开(公告)号:US10262645B1

    公开(公告)日:2019-04-16

    申请号:US15796327

    申请日:2017-10-27

    Inventor: Jonathan S. Abel

    Abstract: In general, the present invention relates to a method and system for synthesizing artificial reverberation using modal analysis of a room or resonating object. In one embodiment of the inventive system, a collection of resonant filters is employed, each driven by the source signal, and their outputs summed. With filter resonance frequencies and dampings tuned to the modal frequencies and decay times of the acoustic space or resonating object being simulated, and filter gains set according to the source and listener positions within the space or object, any number of acoustic spaces and resonant objects may be simulated.

    Method and system for artificial reverberation using modal decomposition

    公开(公告)号:US09805704B1

    公开(公告)日:2017-10-31

    申请号:US14558531

    申请日:2014-12-02

    Inventor: Jonathan S. Abel

    Abstract: In general, the present invention relates to a method and system for synthesizing artificial reverberation using modal analysis of a room or resonating object. In one embodiment of the inventive system, a collection of resonant filters is employed, each driven by the source signal, and their outputs summed. With filter resonance frequencies and dampings tuned to the modal frequencies and decay times of the acoustic space or resonating object being simulated, and filter gains set according to the source and listener positions within the space or object, any number of acoustic spaces and resonant objects may be simulated.

    Three-dimensional virtual audio display employing reduced complexity
imaging filters
    10.
    发明授权
    Three-dimensional virtual audio display employing reduced complexity imaging filters 失效
    采用降低复杂度的成像滤波器的三维虚拟音频显示

    公开(公告)号:US6072877A

    公开(公告)日:2000-06-06

    申请号:US907309

    申请日:1997-08-06

    Inventor: Jonathan S. Abel

    CPC classification number: H04S1/002 H04S1/005 H04S2420/01

    Abstract: A three-dimensional virtual audio display method is described which includes generating a set of transfer function parameters in response to a spatial location or direction signal. An audio signal is filtered in response to the set of transfer function parameters. The set of transfer function parameters are selected from or interpolated among parameters derived by smoothing frequency components of a known transfer function over a bandwidth which is a non-constant function of frequency. The smoothing includes for each frequency component in at least part of the audio band of the display, applying a mean function to the amplitude of the frequency components within the bandwidth containing the frequency component, and noting the parameters of the resulting compressed transfer function.

    Abstract translation: 描述了三维虚拟音频显示方法,其包括响应于空间位置或方向信号生成一组传递函数参数。 响应于一组传递函数参数对音频信号进行滤波。 在通过频率的非常数函数的带宽上通过已知传递函数的平滑频率分量导出的参数中选择或内插传递函数参数的集合。 该平滑包括对于显示器的至少部分音频带中的每个频率分量,对包含频率分量的带宽内的频率分量的振幅应用均值函数,并注意到所得到的压缩传递函数的参数。

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