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22. 发明授权

US09961469B2 Method and device for audio signal processing 有权
公开(公告)号：US09961469B2
公开(公告)日：2018-05-01
申请号：US15022922
申请日：2014-09-17
申请人： WILUS INSTITUTE OF STANDARDS AND TECHNOLOGY INC.
发明人： Taegyu Lee , Hyunoh Oh
IPC分类号： H04S5/00 , H04S3/00 , G10L25/48 , H04S3/02 , G10L19/02 , H03H17/02 , H03H21/00 , H04R5/04 , H04S7/00 , G10L19/008
CPC分类号： H04S5/005 , G10L19/008 , G10L19/0204 , G10L25/48 , H03H17/0229 , H03H17/0248 , H03H17/0266 , H03H17/0272 , H03H21/0012 , H04R5/04 , H04S3/00 , H04S3/002 , H04S3/008 , H04S3/02 , H04S7/30 , H04S2400/01 , H04S2400/03 , H04S2420/03 , H04S2420/07 , H04S2420/11
摘要： The present invention relates to a method and an apparatus for processing a signal, which are used for effectively reproducing an audio signal, and more particularly, to a method and an apparatus for processing a signal, which are used for implementing binaural rendering for reproducing multi-channel or multi-object audio signals in stereo with a low calculation amount.To this end, provided are a method for processing an audio signal including: receiving multi-audio signals including multi-channel or multi-object signals, each of the multi-audio signals including a plurality of subband signals, and the plurality of subband signals including a signal of a first subband group having low frequencies and a signal of a second subband group having high frequencies based on a predetermined frequency band; receiving at least one parameter corresponding to each subband signal of the second subband group, the at least one parameter being extracted from binaural room impulse response (BRIR) subband filter coefficients corresponding to each subband signal of the second subband group; and performing tap-delay line filtering of the subband signal of the second subband group by using the received parameter and an apparatus for processing an audio signal using the same.

23. 发明授权

US09264019B2 Suppression of fixed-pattern jitter using FIR filters 有权
标题翻译：使用FIR滤波器抑制固定模式抖动
公开(公告)号：US09264019B2
公开(公告)日：2016-02-16
申请号：US14321390
申请日：2014-07-01
申请人： ESS Technology, Inc.
发明人： A. Martin Mallinson
IPC分类号： H03H21/00 , H03H17/06 , H03H17/02
CPC分类号： H03H17/06 , H03H17/0229 , H03H17/0248 , H03H17/0286 , H03H17/0294 , H03H21/0012 , H03H21/002
摘要： FIR filters for compensating for fixed pattern jitter, and methods of constructing the same, are disclosed. In one embodiment, a FIR filter filters a signal having a desired frequency component, with the coefficients of the FIR filter selected so that the filter is the equivalent of two combined FIR filters, one having the desired frequency at the filter's peak output frequency, and a second in which the signal is delayed by a time equal to half of a period of a different frequency which is desired to be removed from the output signal. In another embodiment, a FIR filter includes a delay line with a total delay longer than the period of the jitter. A signal is passed down the delay line, the number of signal edges that have occurred as the signal passes each delay element in the counted. Drivers corresponding to the delay elements in which a number of signal edges occur at the desired frequency during the period of fixed pattern jitter activate impedance elements attached to those delay elements. A processor configures the activated impedance elements to provide the desired filter response.
摘要翻译：公开了用于补偿固定模式抖动的FIR滤波器及其构造方法。在一个实施例中，FIR滤波器对具有期望频率分量的信号进行滤波，其中选择FIR滤波器的系数，使得滤波器等效于两个组合的FIR滤波器，一个具有滤波器峰值输出频率处的期望频率，第二个信号被延迟等于期望从输出信号去除的不同频率的周期的一半的时间。在另一个实施例中，FIR滤波器包括具有比抖动周期长的总延迟的延迟线。信号沿着延迟线传递，当信号通过计数的每个延迟元件时，发生的信号边沿的数量。对应于在固定图案抖动的周期期间以期望频率出现的信号边缘数量的延迟元件的驱动器激活附接到这些延迟元件的阻抗元件。处理器配置激活的阻抗元件以提供期望的滤波器响应。

24. 发明申请

US20120308029A1 ADAPTIVE FILTERING SYSTEM 有权
标题翻译：自适应滤波系统
公开(公告)号：US20120308029A1
公开(公告)日：2012-12-06
申请号：US13482678
申请日：2012-05-29
申请人： Markus Christoph
发明人： Markus Christoph
IPC分类号： G10K11/16
CPC分类号： H03H17/0266 , H03H17/0229 , H03H21/0012 , H03H2021/0041
摘要： An audio system with at least one audio channel may include a digital audio processor in which at least one digital filter is implemented for each channel. The digital filter of each channel may include: an analysis filter bank configured to receive a broad-band input audio signal and divide the input audio signal into a plurality of sub-bands, a sub-band filter for each sub-band. and a synthesis filter bank configured to receive the filtered sub-band signals and combine them for providing a broad-band output audio signal. A delay is associated with each sub-band signal, the delay of one of the sub-band signals being applied to the broad-band input audio signal upstream of the analysis filter bank and the residual delays being applied to the remaining sub-band signals downstream of the analysis filter bank.
摘要翻译：具有至少一个音频通道的音频系统可以包括其中为每个通道实现至少一个数字滤波器的数字音频处理器。每个通道的数字滤波器可以包括：分析滤波器组，被配置为接收宽带输入音频信号并将输入音频信号划分成多个子带，每个子带的子带滤波器。以及合成滤波器组，被配置为接收经滤波的子带信号并将其组合以提供宽带输出音频信号。延迟与每个子带信号相关联，一个子带信号的延迟被施加到分析滤波器组上游的宽带输入音频信号，并且剩余延迟被施加到剩余的子带信号分析滤波器组的下游。

25. 发明授权

US07127481B1 Movable tap finite impulse response filter 有权
标题翻译：可动抽头有限脉冲响应滤波器
公开(公告)号：US07127481B1
公开(公告)日：2006-10-24
申请号：US09678728
申请日：2000-10-04
申请人： Yat-Tung Lam
发明人： Yat-Tung Lam
IPC分类号： G00F17/10
CPC分类号： H03H17/0294 , H03H17/0229 , H03H17/06 , H03H21/0012
摘要： Adaptive Finite Impulse Response Filter control includes structure and steps for receiving an input signal, filtering the input signal with an FIR filter having a plurality of filter stages, and delaying application of the input signal to at least one of said filter stages with respect to the other filter stages to skip filtering a portion of the input signal.
摘要翻译：自适应有限脉冲响应滤波器控制包括用于接收输入信号的结构和步骤，用具有多个滤波器级的FIR滤波器对输入信号进行滤波，并且延迟将输入信号施加到至少一个所述滤波级，相对于其他滤波器级跳过过滤输入信号的一部分。

26. 发明申请

US20040143615A1 Finite impulse response filter and digital signal receiving apparatus 有权
标题翻译：有限脉冲响应滤波器和数字信号接收装置
公开(公告)号：US20040143615A1
公开(公告)日：2004-07-22
申请号：US10475090
申请日：2003-10-24
发明人： Hidekuni Yomo , Yoshinori Kunieda , Yuuri Yamamoto
IPC分类号： G06F017/10
CPC分类号： H03H17/0657 , H03H17/0223 , H03H17/0229 , H03H17/06 , H03H17/0621 , H03H2218/085
摘要： An A/D conversion section (101) performs oversampling on an analog signal at a rate M times the symbol rate to convert into digital signals. A FIR filtering section (102) has two delay-element sequences each with a plurality of delay elements, and the two sequences have different delay directions of input signal, i.e., forward direction and reverse direction. The delay directions of input signal can be switched, and according to the finite impulse response train having such delay-element sequences, convolutional calculation is performed. A phase determining section (103) determines a phase used in making a decision in a decision section (104). The decision section (104) makes a decision on a filtered signal using the phase determined in the phase determining section (103) to generate bit data. A digital signal receiving apparatus is thus achieved which determines a phase with high accuracy without increasing the oversampling number, and performs fast calculation while having a reduced circuitry scale.
摘要翻译： A / D转换部（101）对符号率M倍的模拟信号进行过采样，以转换为数字信号。 FIR滤波部分（102）具有两个具有多个延迟元件的延迟元件序列，并且两个序列具有不同的输入信号的延迟方向，即正向和反向。可以切换输入信号的延迟方向，并且根据具有这种延迟元件序列的有限脉冲响应序列，进行卷积计算。相位确定部分（103）确定在判定部分（104）中作出判定所使用的相位。判定部（104）使用在相位确定部（103）中确定的相位对滤波后的信号进行判定，生成位数据。因此实现了一种数字信号接收装置，其在不增加过采样数量的情况下高精度地确定相位，并且在具有减小的电路规模的同时执行快速计算。

27. 发明授权

US5511015A Double-accumulator implementation of the convolution function 失效
标题翻译：卷积函数的双累加器实现
公开(公告)号：US5511015A
公开(公告)日：1996-04-23
申请号：US158757
申请日：1993-11-30
申请人： Stuart W. Flockencier
发明人： Stuart W. Flockencier
IPC分类号： G06F17/15 , H03H17/02 , G06F17/10
CPC分类号： G06F17/15 , H03H17/0229
摘要： A Finite Impulse Response (FIR) digital signal processing circuit uses a double-accumulator technique to drastically reduce the number of multiply-accumulate operations which are necessary per sample of input data. The amount of reduction is dependent upon the shape of the filter function to be convolved. A double-accumulator (D-A) can be implemented by first providing a set of D-A coefficients, which are derived from the filter coefficient stream (FCS). Each D-A coefficient is multiplied by a separate input data sample. The products are summed together along with the result of a previous multiplication of the same D-A coefficients with different input data samples. This first sum is added to another number to form a second sum. The other number is the previous value of the second sum. The second sum is the result.The derived D-A coefficients are fewer and simpler than that required by the conventional FIR implementation. Since multipliers are complex, costly, bulky and limited in speed, the D-A method can lessen these constraints.
摘要翻译：有限脉冲响应（FIR）数字信号处理电路使用双累加技术大大减少了每个输入数据样本所必需的乘法累加次数。减少量取决于要卷积的过滤器功能的形状。可以通过首先提供从滤波器系数流（FCS）导出的一组D-A系数来实现双累加器（D-A）。每个D-A系数乘以单独的输入数据样本。产品与先前乘以相同D-A系数与不同输入数据样本的结果相加在一起。该第一个总和被添加到另一个数字以形成第二个和。另一个数字是第二个和的前一个值。第二个结果是结果。衍生的D-A系数比常规FIR实现所需的更少和更简单。由于乘法器复杂，成本高，体积大，速度有限，因此D-A方法可以减轻这些限制。

28. 发明授权

US09578437B2 Method and apparatus for processing audio signals 有权
标题翻译：用于处理音频信号的方法和装置
公开(公告)号：US09578437B2
公开(公告)日：2017-02-21
申请号：US14990814
申请日：2016-01-08
申请人： WILUS INSTITUTE OF STANDARDS AND TECHNOLOGY INC.
发明人： Hyunoh Oh , Taegyu Lee
IPC分类号： H04R5/02 , H04S5/00 , H04S3/00 , G10L19/008 , H03H17/02 , H04R5/04 , H04S7/00
CPC分类号： H04S5/005 , G10L19/008 , G10L19/0204 , G10L25/48 , H03H17/0229 , H03H17/0248 , H03H17/0266 , H03H17/0272 , H03H21/0012 , H04R5/04 , H04S3/00 , H04S3/002 , H04S3/008 , H04S3/02 , H04S7/30 , H04S2400/01 , H04S2400/03 , H04S2420/03 , H04S2420/07 , H04S2420/11
摘要： The present invention relates to a method and an apparatus for processing a signal, which are used for effectively reproducing an audio signal, and more particularly, to a method and an apparatus for processing a signal, which are used for implementing binaural rendering for reproducing multi-channel or multi-object audio signals in stereo with a low calculation amount.To this end, provided are a method for processing an audio signal including: receiving multi-audio signals including multi-channel or multi-object signals; receiving truncated subband filter coefficients for filtering the multi-audio signals, the truncated subband filter coefficients being at least a portion of subband filter coefficients obtained from a binaural room impulse response (BRIR) filter coefficient for binaural filtering of the multi-audio signals, and the lengths of the truncated subband filter coefficients being determined based on filter order information obtained by at least partially using characteristic information extracted from the corresponding subband filter coefficients; and filtering the subband signal by using the truncated subband filter coefficients corresponding to each subband signal of the multi-audio signals and an apparatus for processing an audio signal using the same.
摘要翻译：为此，提供了一种用于处理音频信号的方法，包括：接收包括多声道或多对象信号的多音频信号; 接收用于对多音频信号进行滤波的截断的子带滤波器系数，截取的子带滤波器系数是从用于多音频信号的双耳滤波的双耳室脉冲响应（BRIR）滤波器系数获得的子带滤波器系数的至少一部分，以及基于通过至少部分地使用从相应的子带滤波器系数提取的特征信息获得的滤波器顺序信息来确定截断的子带滤波器系数的长度; 以及通过使用与多音频信号的每个子带信号相对应的截断的子带滤波器系数和使用其来处理音频信号的装置来对子带信号进行滤波。

29. 发明申请

US20160234620A1 METHOD AND DEVICE FOR AUDIO SIGNAL PROCESSING 有权
标题翻译：用于音频信号处理的方法和装置
公开(公告)号：US20160234620A1
公开(公告)日：2016-08-11
申请号：US15022922
申请日：2014-09-17
申请人： WILUS INSTITUTE OF STANDARDS AND TECHNOLOGY INC.
发明人： Taegyu LEE , Hyunoh OH
IPC分类号： H04S7/00 , H04R5/04 , H03H17/02 , G10L19/008
CPC分类号： H04S5/005 , G10L19/008 , G10L19/0204 , G10L25/48 , H03H17/0229 , H03H17/0248 , H03H17/0266 , H03H17/0272 , H03H21/0012 , H04R5/04 , H04S3/00 , H04S3/002 , H04S3/008 , H04S3/02 , H04S7/30 , H04S2400/01 , H04S2400/03 , H04S2420/03 , H04S2420/07 , H04S2420/11
摘要： The present invention relates to a method and an apparatus for processing a signal, which are used for effectively reproducing an audio signal, and more particularly, to a method and an apparatus for processing a signal, which are used for implementing binaural rendering for reproducing multi-channel or multi-object audio signals in stereo with a low calculation amount.To this end, provided are a method for processing an audio signal including: receiving multi-audio signals including multi-channel or multi-object signals, each of the multi-audio signals including a plurality of subband signals, and the plurality of subband signals including a signal of a first subband group having low frequencies and a signal of a second subband group having high frequencies based on a predetermined frequency band; receiving at least one parameter corresponding to each subband signal of the second subband group, the at least one parameter being extracted from binaural room impulse response (BRIR) subband filter coefficients corresponding to each subband signal of the second subband group; and performing tap-delay line filtering of the subband signal of the second subband group by using the received parameter and an apparatus for processing an audio signal using the same.
摘要翻译：本发明涉及一种用于处理信号的方法和装置，其用于有效地再现音频信号，更具体地，涉及一种用于处理信号的方法和装置，其用于实现用于再现多声道的双耳渲染通道或多对象音频信号，具有低计算量的立体声。为此，提供了一种处理音频信号的方法，包括：接收包括多声道或多对象信号的多音频信号，每个多音频信号包括多个子带信号，以及多个子带信号包括基于预定频带的具有低频的第一子带组的信号和具有高频率的第二子带组的信号; 接收与所述第二子带组的每个子带信号对应的至少一个参数，所述至少一个参数是从与所述第二子带组的每个子带信号相对应的双耳室脉冲响应（BRIR）子带滤波器系数中提取的; 以及通过使用所接收的参数来执行第二子带组的子带信号的抽头延迟线滤波，以及使用该参数来处理音频信号的装置。

30. 发明申请

US20160198281A1 METHOD AND APPARATUS FOR PROCESSING AUDIO SIGNALS 有权
标题翻译：处理音频信号的方法和装置
公开(公告)号：US20160198281A1
公开(公告)日：2016-07-07
申请号：US14990814
申请日：2016-01-08
申请人： WILUS INSTITUTE OF STANDARDS AND TECHNOLOGY INC.
发明人： Hyunoh OH , Taegyu LEE
IPC分类号： H04S5/00 , G10L19/008
CPC分类号： H04S5/005 , G10L19/008 , G10L19/0204 , G10L25/48 , H03H17/0229 , H03H17/0248 , H03H17/0266 , H03H17/0272 , H03H21/0012 , H04R5/04 , H04S3/00 , H04S3/002 , H04S3/008 , H04S3/02 , H04S7/30 , H04S2400/01 , H04S2400/03 , H04S2420/03 , H04S2420/07 , H04S2420/11
摘要： The present invention relates to a method and an apparatus for processing a signal, which are used for effectively reproducing an audio signal, and more particularly, to a method and an apparatus for processing a signal, which are used for implementing binaural rendering for reproducing multi-channel or multi-object audio signals in stereo with a low calculation amount.To this end, provided are a method for processing an audio signal including: receiving multi-audio signals including multi-channel or multi-object signals; receiving truncated subband filter coefficients for filtering the multi-audio signals, the truncated subband filter coefficients being at least a portion of subband filter coefficients obtained from a binaural room impulse response (BRIR) filter coefficient for binaural filtering of the multi-audio signals, and the lengths of the truncated subband filter coefficients being determined based on filter order information obtained by at least partially using characteristic information extracted from the corresponding subband filter coefficients; and filtering the subband signal by using the truncated subband filter coefficients corresponding to each subband signal of the multi-audio signals and an apparatus for processing an audio signal using the same.
摘要翻译：本发明涉及一种用于处理信号的方法和装置，其用于有效地再现音频信号，更具体地，涉及一种用于处理信号的方法和装置，其用于实现用于再现多声道的双耳渲染通道或多对象音频信号，具有低计算量的立体声。为此，提供了一种用于处理音频信号的方法，包括：接收包括多声道或多对象信号的多音频信号; 接收用于对多音频信号进行滤波的截断的子带滤波器系数，截取的子带滤波器系数是从用于多音频信号的双耳滤波的双耳室脉冲响应（BRIR）滤波器系数获得的子带滤波器系数的至少一部分，以及基于通过至少部分地使用从相应的子带滤波器系数提取的特征信息获得的滤波器顺序信息来确定截断的子带滤波器系数的长度; 以及通过使用与多音频信号的每个子带信号相对应的截断的子带滤波器系数和使用其来处理音频信号的装置来对子带信号进行滤波。

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