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1. 发明授权

US08149156B1 System and method for estimating location of projectile source or shooter location 有权
标题翻译：用于估计射弹源或射手位置的位置的系统和方法
公开(公告)号：US08149156B1
公开(公告)日：2012-04-03
申请号：US12468620
申请日：2009-05-19
申请人： Rustin W. Allred , Robert R. Bless
发明人： Rustin W. Allred , Robert R. Bless
IPC分类号： G01S13/86
CPC分类号： G01S13/86 , F41G3/147 , G01S3/802 , G01S13/583 , G01S13/68 , G01S13/88
摘要： A system for estimating a location of the source of a projectile includes a radar system for transmitting a wave and detecting and providing an indication of a wave reflection from the projectile, an acoustic detection system for detecting and providing an indication of at least one sound associated with the projectile, and circuitry for estimating the location in response to the indication of a wave reflection and the indication of at least one sound.
摘要翻译：用于估计射弹源的位置的系统包括用于发射波的雷达系统，并且检测并提供来自射弹的波反射的指示;声检测系统，用于检测并提供至少一个声音相关联的指示以及用于响应于波反射的指示和至少一个声音的指示来估计位置的电路。

2. 发明授权

US07664808B2 Efficient real-time computation of FIR filter coefficients 有权
标题翻译： FIR滤波系数的有效实时计算
公开(公告)号：US07664808B2
公开(公告)日：2010-02-16
申请号：US11159618
申请日：2005-06-23
申请人： Zhongnong Jiang , Rustin W. Allred
发明人： Zhongnong Jiang , Rustin W. Allred
IPC分类号： G06F17/10
CPC分类号： H03H17/0607
摘要： Systems and methods for determining coefficients of an Finite Impulse Response (FIR) filter are disclosed. The FIR filter coefficients are computed by determining a sine of an input value and an inverse of the input value. The sine of the input signal and the inverse of the input signal are multiplied together to form a sinc value of the input value. The sinc value is employed to determine the coefficient. The system and method can be repeated to compute any number of FIR filter coefficients in real-time. The sine of the input signal is computed utilizing a memory lookup table. The memory lookup table includes pairs of uniformly distributed values for the sine and cosine functions in the range of 0 to π. The inverse of the input value is computed using an inverse memory lookup table, a most significant digit and a remainder. The coefficient is then computed from a product of the sine of the input signal and the inverse of the input signal. Thus, the coefficient is computable in real-time without the use of previously computed and stored coefficients.
摘要翻译：公开了用于确定有限脉冲响应（FIR）滤波器的系数的系统和方法。通过确定输入值的正弦和输入值的倒数来计算FIR滤波器系数。将输入信号的正弦和输入信号的反相相乘，形成输入值的sinc值。采用sinc值来确定系数。可以重复系统和方法来实时计算任意数量的FIR滤波器系数。使用存储器查找表来计算输入信号的正弦。存储器查找表包括在0到pi的范围内的正弦和余弦函数的均匀分布值对。使用逆存储器查找表，最高有效数字和余数计算输入值的倒数。然后根据输入信号的正弦和输入信号的反相的乘积计算系数。因此，该系数可以实时计算，而不使用先前计算和存储的系数。

3. 发明授权

US06965676B1 Volume-responsive loudness compensation circuits, systems, and methods 有权
标题翻译：音量响应响度补偿电路，系统和方法
公开(公告)号：US06965676B1
公开(公告)日：2005-11-15
申请号：US09603950
申请日：2000-06-27
申请人： Rustin W Allred
发明人： Rustin W Allred
IPC分类号： H04R3/04 , H03G3/00 , H03G5/00 , H03G9/00 , H03H17/00
CPC分类号： H03G5/005
摘要： An audio compensation system (20) for producing a sound compensated output signal (So). The system comprises a volume control circuit (24) for producing a volume-adjusted signal (Sic) by applying a volume adjustment to an audio signal in response to a volume setting, wherein the sound compensated output signal is responsive to the volume-adjusted signal. The system also comprises circuitry (26, 30, 32) for producing an amplified signal (Sifa) by amplifying a selected bandwith of signals in response to the volume setting. The sound compensated output signal is also responsive to the amplified signal.
摘要翻译：一种用于产生声音补偿输出信号（S O）的音频补偿系统（20）。该系统包括用于响应于音量设置对音频信号进行音量调节来产生音量调节信号的音量控制电路（24），其中声音补偿的输出信号响应于音量调节信号。该系统还包括用于通过响应于音量设置放大所选频带的信号来产生放大的信号（S）的电路（26,30,32）。声音补偿输出信号也响应于放大的信号。

4. 发明授权

US06892103B1 Digital tone control with linear step coefficients 有权
标题翻译：具有线性步进系数的数字音调控制
公开(公告)号：US06892103B1
公开(公告)日：2005-05-10
申请号：US09408095
申请日：1999-09-27
申请人： Rustin W. Allred , Robert S. Young, Jr. , Michael J. Tsecouras
发明人： Rustin W. Allred , Robert S. Young, Jr. , Michael J. Tsecouras
IPC分类号： H03G5/00
CPC分类号： H03G5/005
摘要： Digital audio tone control implemented using Shelving filters for the digital audio treble tone control exhibits artifacts (noise, distortion, etc.) as the tone control settings are changed. This was previously accomplished by changing filter coefficients in the traditional small equal (on a dB scale) filter steps of a fraction of 1 dB. While this worked for bass filters, artifacts were still present for treble. This invention eliminates these artifacts by changing the filter steps to small equal steps on a linear scale. Additionally, where the steps became too large for the resolution required, additional filter steps are added. Approximately 150 filter steps are used for treble control and 128 filter steps are used for bass tone control. An efficient way of implementing the filter steps for digital tone control stores (119) one set of filter coefficient values and a small amount of additional information and then increments the coefficients between all the other steps. This reduces the memory required by as much as 95% or the machine cycles for implementing filter coefficients by 40-200 times. This new efficient method is accomplished by: 1) defining the filter coefficients, 2) piecewise linearizing the plot of filter coefficients versus filter step to define the linear regions, 3) defining the initial set of filter coefficients, and 4) defining increments between the filter steps.
摘要翻译：数字音频音调控制使用数字音频高音音调控制的搁架滤波器，随着音调控制设置的改变，会出现伪像（噪音，失真等）。以前通过改变传统的等于（在dB尺度上）小于1dB的滤波器步长的滤波器系数来实现。虽然这对低音滤波器有效，但仍然存在高音的文物。本发明通过将滤波器步骤改变为在线性尺度上的小的相等步骤来消除这些伪影。此外，如果步骤变得太大以致于需要解析，则会添加额外的过滤器步骤。大约150个滤波器步骤用于高音控制，128个滤波器步骤用于低音控制。实现数字音调控制存储器（119）的滤波器步骤的一种有效方法是一组滤波器系数值和少量附加信息，然后在所有其它步骤之间增加系数。这将所需的存储量减少了95％，或者将滤波器系数实现40-200次的机器周期。这种新的有效方法是通过以下方式实现的：1）定义滤波器系数，2）将滤波器系数的曲线与滤波器步长分段线性化以定义线性区域，3）定义初始滤波器系数集合，以及4）过滤步骤

5. 发明授权

US06834292B2 Multi-rate digital filter for audio sample-rate conversion 有权
标题翻译：用于音频采样率转换的多速率数字滤波器
公开(公告)号：US06834292B2
公开(公告)日：2004-12-21
申请号：US10219145
申请日：2002-08-15
申请人： Zhongnong Jiang , Rustin W. Allred , James R. Hochschild
发明人： Zhongnong Jiang , Rustin W. Allred , James R. Hochschild
IPC分类号： G06F1717
CPC分类号： H03H17/0275 , H03H17/0685
摘要： In a microprocessor, a method for providing a sample-rate conversion (“SRC”) filter on an input stream of sampled data provided at a first rate, to produce an output stream of data at a second rate different from the first rate. The input stream of sampled data is operated on with a first low-order interpolation filter routine to produce a first stream of intermediate data. The first stream of intermediate data is operated on with a first simplified interpolation filter routine, having a substantially small number of operations to calculate the coefficients thereof, to produce a second stream of intermediate data. The second stream of intermediate data is operated on with a first decimating filter routine to produce the output stream of data.
摘要翻译：在微处理器中，一种用于在以第一速率提供的采样数据的输入流上提供采样率转换（“SRC”）滤波器的方法，以产生与第一速率不同的第二速率的输出数据流。采样数据的输入流利用第一低阶内插滤波程序进行操作，以产生第一中间数据流。中间数据的第一个流是利用第一简化内插滤波程序进行操作的，其具有基本上少量的操作以计算其系数，以产生第二中间数据流。第二个中间数据流通过第一抽取滤波器程序进行操作，以产生输出的数据流。

6. 发明授权

US07191025B2 Variable digital high and low pass filters 有权
标题翻译：可变数字高通滤波器和低通滤波器
公开(公告)号：US07191025B2
公开(公告)日：2007-03-13
申请号：US10326517
申请日：2002-12-20
申请人： Rustin W. Allred
发明人： Rustin W. Allred
IPC分类号： G06F17/00 , G06F17/10
CPC分类号： H03H17/0294 , H03G5/005
摘要： A system (10) for providing an integer number N of filters. The system comprises an input (Di) for receiving a digital audio signal and an output (Do) for providing a filtered audio signal. The system also comprises circuitry (16) for storing at least a first set of fixed filter coefficients and circuitry for storing estimation data. The system also comprises circuitry (14) for estimating a number of sets of estimated filter coefficients in response to the estimation data and the fixed filter coefficients. The system also comprises circuitry (14) for applying a transfer function to the digital audio signal and in response for providing the filtered audio signal. The circuitry for applying the transfer function applies a set of filter coefficients selected from the first set of fixed filter coefficients and the sets of estimated filter coefficients. Also, the transfer function is selected from a transfer function set consisting of a high pass filter transfer function and a low pass filter transfer function.
摘要翻译：一种用于提供整数N个滤波器的系统（10）。该系统包括用于接收数字音频信号的输入端（D SUB）和用于提供经滤波的音频信号的输出端（D SUB）。该系统还包括用于存储至少第一组固定滤波器系数的电路（16）和用于存储估计数据的电路。该系统还包括用于响应于估计数据和固定滤波器系数来估计估计的滤波器系数的集合的数量的电路（14）。该系统还包括用于将传递函数应用于数字音频信号并且响应于提供经滤波的音频信号的电路（14）。用于应用传递函数的电路应用从第一组固定滤波器系数和估计的滤波器系数集合中选择的一组滤波器系数。此外，传递函数从由高通滤波器传递函数和低通滤波器传递函数组成的传递函数集中选择。

7. 发明授权

US07107300B2 Circuits, systems, and methods implementing approximations for inverse logarithm 有权
标题翻译：电路，系统和实现反对数近似的方法
公开(公告)号：US07107300B2
公开(公告)日：2006-09-12
申请号：US10150723
申请日：2002-05-17
申请人： Rustin W. Allred
发明人： Rustin W. Allred
IPC分类号： G06F1/02
CPC分类号： G06F7/556
摘要： A digital signal system (50) determines an approximate antilog x from a value of ƒ(x), where x has a base b.
摘要翻译：数字信号系统（50）从f（x）的值确定近似的反对数x，其中x具有基数b。

8. 发明授权

US07058188B1 Configurable digital loudness compensation system and method 有权
标题翻译：可配置的数字响度补偿系统及方法
公开(公告)号：US07058188B1
公开(公告)日：2006-06-06
申请号：US09421417
申请日：1999-10-19
申请人： Rustin W. Allred
发明人： Rustin W. Allred
IPC分类号： H03G3/00 , H03G5/00
CPC分类号： H03G5/18
摘要： An audio loudness compensation system includes a level sensor receiving an audio input signal and operable to estimate a level of the audio input signal over a first predetermined time period, and a level mapper receiving the estimated level and operable to map the estimated level to a raw audio gain in response to a slope setting and an offset setting. The system further includes an attack and decay filter receiving the raw audio gain and operable to smooth out increasing and decreasing changes in the raw audio gain in response to a second and, possibly a third predetermined time period, and a compensation filter receiving the smoothed raw audio gain and operable to modify the audio input signal in response to the smoothed raw audio gain, a center frequency setting and a bandwidth setting, and generate a loudness compensated audio output signal.
摘要翻译：音频响度补偿系统包括接收音频输入信号并可操作以在第一预定时间段内估计音频输入信号的电平的电平传感器和接收估计电平的电平映射器，并可操作以将估计电平映射到原始响应于斜率设置和偏移设置的音频增益。该系统还包括接收原始音频增益的攻击和衰减滤波器，并且可操作以响应于第二预定时间段（可能第三预定时间周期）平滑原始音频增益的增加和减小的改变，以及接收平滑的原始音频增益，并且可操作以响应于平滑的原始音频增益，中心频率设置和带宽设置来修改音频输入信号，并且生成响度补偿音频输出信号。

9. 发明授权

US06487573B1 Multi-rate digital filter for audio sample-rate conversion 有权
标题翻译：用于音频采样率转换的多速率数字滤波器
公开(公告)号：US06487573B1
公开(公告)日：2002-11-26
申请号：US09277696
申请日：1999-03-26
申请人： Zhongnong Jiang , Rustin W. Allred , James R. Hochschild
发明人： Zhongnong Jiang , Rustin W. Allred , James R. Hochschild
IPC分类号： G06F1717
CPC分类号： H03H17/0275 , H03H17/0685
摘要： A method for providing a sample-rate conversion (“SRC”) filter on an input stream of sampled data provided at a first rate, to produce an output stream of data at a second rate different from the first rate. The input stream of sampled data is operated on with a first low-order interpolation filter routine to produce a first stream of intermediate data. The first stream of intermediate data is operated on with a first simplified interpolation filter routine, having a substantially small number of operations to calculate the coefficients thereof, to produce a second stream of intermediate data. The second stream of intermediate data is operated on with a first decimating filter routine to produce the output stream of data.
摘要翻译：一种用于在以第一速率提供的采样数据的输入流上提供采样率转换（“SRC”）滤波器的方法，以产生与第一速率不同的第二速率的输出数据流。采样数据的输入流利用第一低阶内插滤波程序进行操作，以产生第一中间数据流。中间数据的第一个流是利用第一简化内插滤波程序进行操作的，其具有基本上少量的操作以计算其系数，以产生第二中间数据流。第二个中间数据流通过第一抽取滤波器程序进行操作，以产生输出的数据流。

10. 发明授权

US07058126B1 Digital graphametric equalizer 有权
标题翻译：数字图形均衡器
公开(公告)号：US07058126B1
公开(公告)日：2006-06-06
申请号：US09481851
申请日：2000-01-14
申请人： Rustin W. Allred
发明人： Rustin W. Allred
IPC分类号： H03H7/40
CPC分类号： H03G5/005
摘要： A graphametric equalizer has graphic and parametric equalization capabilities within a single non-redundant system. A translation function capability converts user selected inputs for center frequency, bandwidth and gain into allpass filter parameters to realize an allpass filter-based equalization filter structure capable of performing graphic and/or parametric equalization on-the-fly. The graphametric equalizer has a softening function capability to time user inputs and increment filter parameters gracefully such that the graphametric equalizer can be recharacterized with new filter parameters on-the-fly without incurring undesirable audible artifacts.
摘要翻译：图形均衡器在单个非冗余系统中具有图形和参数均衡功能。翻译功能能力将用户选择的中心频率，带宽和增益输入转换为全通滤波器参数，以实现能够在运行中执行图形和/或参数均衡的基于全通滤波器的均衡滤波器结构。图形均衡器具有软化功能，可以对用户输入进行优化，并优化递增滤波器参数，使得图形均衡器可以在新的滤波参数上实时重新定位，而不会产生不希望的可听见的伪像。

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