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1. 发明授权

US09084049B2 Automatic equalization using adaptive frequency-domain filtering and dynamic fast convolution 有权
标题翻译：使用自适应频域滤波和动态快速卷积自动均衡
公开(公告)号：US09084049B2
公开(公告)日：2015-07-14
申请号：US13878705
申请日：2011-09-13
申请人： Louis D. Fielder , David S. McGrath
发明人： Louis D. Fielder , David S. McGrath
IPC分类号： H03G5/00 , H03B29/00 , H04R3/04 , H03H17/06 , H04L25/03
CPC分类号： H04R3/04 , H03H17/06 , H04L25/03286
摘要： Frequency-domain techniques are used for adaptive equalization that is responsive to spectral magnitude characteristics but not sensitive to phase characteristics of system response. Signal correlation may be used to improve adaptation accuracy when significant levels of ambient sounds are present. A preferred filter implementation uses convolution-based block transforms and cross-fade windows.
摘要翻译：频域技术用于响应于光谱幅度特性但对系统响应的相位特性不敏感的自适应均衡。当存在显着水平的环境声音时，可以使用信号相关性来提高适应精度。优选的滤波器实现使用基于卷积的块变换和交叉渐变窗口。

2. 发明申请

US20130208917A1 Automatic Equalization Using Adaptive Frequency-Domain Filtering and Dynamic Fast Convolution 有权
标题翻译：使用自适应频域滤波和动态快速卷积的自动均衡
公开(公告)号：US20130208917A1
公开(公告)日：2013-08-15
申请号：US13878705
申请日：2011-09-13
申请人： Louis D. Fielder , David S. McGrath
发明人： Louis D. Fielder , David S. McGrath
IPC分类号： H04R3/04
CPC分类号： H04R3/04 , H03H17/06 , H04L25/03286
摘要： Frequency-domain techniques are used for adaptive equalization that is responsive to spectral magnitude characteristics but not sensitive to phase characteristics of system response. Signal correlation may be used to improve adaptation accuracy when significant levels of ambient sounds are present. A preferred filter implementation uses convolution-based block transforms and cross-fade windows.
摘要翻译：频域技术用于响应于光谱幅度特性但对系统响应的相位特性不敏感的自适应均衡。当存在显着水平的环境声音时，可以使用信号相关性来提高适应精度。优选的滤波器实现使用基于卷积的块变换和交叉渐变窗口。

3. 发明授权

US07634092B2 Head related transfer functions for panned stereo audio content 有权
公开(公告)号：US07634092B2
公开(公告)日：2009-12-15
申请号：US10965130
申请日：2004-10-14
申请人： David S. McGrath
发明人： David S. McGrath
IPC分类号： H04R5/00 , H04R1/10 , H04R5/02
CPC分类号： H04S3/00 , H04S2400/01 , H04S2420/01
摘要： A method to process audio signals, an apparatus accepting audio signals, a carrier medium that carried instructions for a processor to implement the method to process audio signals, and a carrier medium carrying filter data to implement a filter of audio signals. The method includes filtering a pair of audio input signals by a process that produces a pair of output signals corresponding to the results of: filtering each of the input signals with a HRTF filter pair, and adding the HRTF filtered signals. The HRTF filter pair is such that a listener listening to the pair of output signals through headphones experiences sounds from a pair of desired virtual speaker locations. Furthermore, the filtering is such that, in the case that the pair of audio input signals includes a panned signal component, the listener listening to the pair of output signals through headphones is provided with the sensation that the panned signal component emanates from a virtual sound source at a center location between the virtual speaker locations.

4. 发明申请

US20120308038A1 Sound Source Localization Apparatus and Method 有权
标题翻译：声源定位装置及方法
公开(公告)号：US20120308038A1
公开(公告)日：2012-12-06
申请号：US13469587
申请日：2012-05-11
申请人： Zhiwei Shuang , David S. McGrath , Glenn N. Dickins
发明人： Zhiwei Shuang , David S. McGrath , Glenn N. Dickins
IPC分类号： H04R3/00
CPC分类号： G01S3/8034
摘要： Sound source localization apparatuses and methods are described. A frame amplitude difference vector is calculated based on short time frame data acquired through an array of microphones. The frame amplitude difference vector reflects differences between amplitudes captured by microphones of the array during recording the short time frame data. Similarity between the frame amplitude difference vector and each of a plurality of reference frame amplitude difference vectors is evaluated. Each of the plurality of reference frame amplitude difference vectors reflects differences between amplitudes captured by microphones of the array during recording sound from one of a plurality of candidate locations. A desired location of sound source is estimated based at least on the candidate locations and associated similarity. The sound source localization can be performed based at least on amplitude difference.
摘要翻译：描述声源定位装置和方法。基于通过麦克风阵列获取的短时帧数据来计算帧幅度差矢量。帧幅度差矢量反映在记录短时帧数据期间由阵列的麦克风捕获的幅度之间的差异。评估帧幅度差矢量与多个参考帧幅度差矢量中的每一个之间的相似度。多个参考帧幅度差矢量中的每一个在从多个候选位置之一记录声音期间反映阵列的麦克风捕获的幅度之间的差异。至少基于候选位置和相关联的相似性来估计声源的期望位置。可以至少基于幅度差进行声源定位。

5. 发明申请

US20120128159A1 Decorrelator for Upmixing Systems 有权
标题翻译：上混系统的相关器
公开(公告)号：US20120128159A1
公开(公告)日：2012-05-24
申请号：US13121323
申请日：2009-09-28
申请人： David S. McGrath , Mark S. Vinton
发明人： David S. McGrath , Mark S. Vinton
IPC分类号： H04R5/00
CPC分类号： H04S1/002 , H04S3/002 , H04S7/307
摘要： An improved decorrelator is disclosed that processes an input audio signal in two separate paths. In one path, a banded phase-flip filter is applied to lower frequencies of the input audio signal. In a second path, a frequency-dependent delay is applied to higher frequencies of the input audio signal. Signals from the two paths are combined to obtain an output signal that is psychoacoustically decorrelated with the input audio signal. The decorrelated signal can be mixed with the input audio signal without generating audible artifacts.
摘要翻译：公开了一种改进的去相关器，其处理两个独立路径中的输入音频信号。在一个路径中，带状相位倒相滤波器被施加到输入音频信号的较低频率。在第二路径中，频率依赖的延迟被施加到输入音频信号的较高频率。来自两个路径的信号被组合以获得与输入音频信号心理声学去相关的输出信号。解相关信号可以与输入音频信号混合，而不会产生可听见的伪影。

6. 发明授权

US06847920B2 Approximation sequence processing 有权
标题翻译：近似序列处理
公开(公告)号：US06847920B2
公开(公告)日：2005-01-25
申请号：US10630356
申请日：2003-07-30
申请人： David S. McGrath
发明人： David S. McGrath
IPC分类号： G06F17/17 , H03M3/00 , H04B1/00
CPC分类号： G06F17/17 , H03M3/478
摘要： A method of producing an approximation sequence to a series of sample values, the method comprising the steps of (a) determining a first set having candidate partial sequences as members, each member comprising a plurality of elements; (b) selecting the first n elements of one of the members of the first set as a next output element for said approximation sequence; n a positive integer; (c) forming a second set having descendent candidate partial sequences as members from said first set; (d) applying a fitness filtering process to said second set to rank its members according to fitness for representing at least a corresponding portion of the series of input samples; (e) selecting at least some of the members of the second set to form a third set; and repeating steps (a) to (e) so as to produce said approximation sequence, wherein the third set of step (e) functions as the first set of the subsequent step (a).
摘要翻译：一种对一系列样本值产生近似序列的方法，所述方法包括以下步骤：（a）确定具有候选部分序列作为成员的第一集合，每个部件包括多个元素; （b）选择第一组成员之一的前n个元素作为所述近似序列的下一个输出元素; n为正整数; （c）形成具有来自所述第一组的成员的后代候选部分序列的第二集合; （d）对所述第二集合应用适应度滤波处理，以根据适于表示所述一系列输入样本的至少相应部分的适合度对其成员进行排序; （e）选择第二组的至少一些成员以形成第三组; 以及重复步骤（a）至（e）以产生所述近似序列，其中步骤（e）的第三组用作随后步骤（a）的第一组。

7. 发明授权

US5502747A Method and apparatus for filtering an electronic environment with improved accuracy and efficiency and short flow-through delay 失效
标题翻译：用于以提高的精度和效率以及短的流通延迟来过滤电子环境的方法和装置
公开(公告)号：US5502747A
公开(公告)日：1996-03-26
申请号：US87125
申请日：1993-07-07
申请人： David S. McGrath
发明人： David S. McGrath
IPC分类号： G06F17/10 , G06F17/14 , H03H17/02 , H03H17/06 , H03H17/08 , H04B1/10
CPC分类号： H03H17/06 , G06F17/142 , H03H17/0213
摘要： An improved method and apparatus for filtering an electronic environment with relatively high accuracy and efficiency and relatively short flow-through delay ("latency") is disclosed. Embodiments of the invention may be applied to digital filters implemented in software, hardware or a combination of both for applications such as audio filtering or electronic modelling of acoustic system characteristics. The method disclosed is broadly applicable in the field of signal processing and may be used to advantage, for example, in adaptive filtering; audio reverberation processing; adaptive echo cancellation; spatial processing; virtual reality audio; correlation, radar; radar pulse compression; deconvolution; seismic analysis; telecommunications; pattern recognition; robotics; 3D acoustic modelling; audio post production (including auralization and auto reverberant matching); audio equalization; compression; sonar; ultrasonics; secure communication systems; digital audio broadcast, acoustic analysis, surveillance; noise cancellation; and echo cancellation.
摘要翻译：公开了一种用于以比较高的精度和效率以及较短的流通延迟（“延迟”）来过滤电子环境的改进的方法和装置。本发明的实施例可以应用于以软件，硬件或两者的组合实现的数字滤波器，用于例如音频滤波或声学系统特性的电子建模。所公开的方法广泛地应用于信号处理领域，并且可以用于例如自适应滤波中的优点; 音频混响处理; 自适应回波消除; 空间处理; 虚拟现实音频; 相关性，雷达; 雷达脉冲压缩; 去卷积地震分析; 电信; 模式识别; 机器人 3D声学建模; 音频后期制作（包括音响和自动混响匹配）; 音频均衡; 压缩; 声纳超声波安全通信系统; 数字音频广播，声学分析，监控; 噪音消除; 和回声消除。

8. 发明授权

US09229086B2 Sound source localization apparatus and method 有权
标题翻译：声源定位装置及方法
公开(公告)号：US09229086B2
公开(公告)日：2016-01-05
申请号：US13469587
申请日：2012-05-11
申请人： Zhiwei Shuang , David S. McGrath , Glenn N. Dickins
发明人： Zhiwei Shuang , David S. McGrath , Glenn N. Dickins
IPC分类号： G01S3/803 , H04R3/00
CPC分类号： G01S3/8034
摘要： Sound source localization apparatuses and methods are described. A frame amplitude difference vector is calculated based on short time frame data acquired through an array of microphones. The frame amplitude difference vector reflects differences between amplitudes captured by microphones of the array during recording the short time frame data. Similarity between the frame amplitude difference vector and each of a plurality of reference frame amplitude difference vectors is evaluated. Each of the plurality of reference frame amplitude difference vectors reflects differences between amplitudes captured by microphones of the array during recording sound from one of a plurality of candidate locations. A desired location of sound source is estimated based at least on the candidate locations and associated similarity. The sound source localization can be performed based at least on amplitude difference.
摘要翻译：描述声源定位装置和方法。基于通过麦克风阵列获取的短时帧数据来计算帧幅度差矢量。帧幅度差矢量反映在记录短时帧数据期间由阵列的麦克风捕获的幅度之间的差异。评估帧幅度差矢量与多个参考帧幅度差矢量中的每一个之间的相似度。多个参考帧幅度差矢量中的每一个在从多个候选位置之一记录声音期间反映阵列的麦克风捕获的幅度之间的差异。至少基于候选位置和相关联的相似性来估计声源的期望位置。可以至少基于幅度差进行声源定位。

9. 发明授权

US09173048B2 Method and system for generating a matrix-encoded two-channel audio signal 有权
标题翻译：用于生成矩阵编码的双声道音频信号的方法和系统
公开(公告)号：US09173048B2
公开(公告)日：2015-10-27
申请号：US14239510
申请日：2012-08-14
申请人： David S. McGrath
发明人： David S. McGrath
IPC分类号： H04S3/02
CPC分类号： H04S3/02
摘要： In some embodiments, a method for generating a matrix-encoded two-channel audio signal in response to a horizontal B-format signal by performing a mixing operation. In other embodiments, a method for generating a matrix-encoded two-channel audio signal, including steps of generating microphone output signals (by capturing sound with a microphone array), and performing a mixing operation on the microphone output signals, where the mixing operation is equivalent to generating a horizontal B-format signal in response to the microphone output signals, and generating the matrix-encoded two-channel audio signal in response to the horizontal B-format signal. The microphone array is typically a small array of cardiod microphones (e.g., an array consisting of three cardiod microphones). Other aspects include systems (e.g., encoders) programmed or otherwise configured to perform any embodiment of the method for generating a matrix-encoded two-channel audio signal.
摘要翻译：在一些实施例中，一种用于通过执行混合操作来响应于水平B格式信号来产生矩阵编码的双声道音频信号的方法。在其他实施例中，一种用于产生矩阵编码的双声道音频信号的方法，包括产生麦克风输出信号（通过用麦克风阵列捕获声音）和对麦克风输出信号执行混频操作的步骤，其中混合操作等效于响应于麦克风输出信号产生水平B格式信号，并且响应于水平B格式信号产生矩阵编码的双声道音频信号。麦克风阵列通常是少量的心形麦克风（例如，由三个心形麦克风组成的阵列）。其它方面包括已编程或以其他方式配置为执行用于生成矩阵编码的双声道音频信号的方法的任何实施例的系统（例如，编码器）。

10. 发明申请

US20120170769A1 Adjusting the Loudness of an Audio Signal with Perceived Spectral Balance Preservation 有权
标题翻译：调整感知光谱平衡保存音频信号的响度
公开(公告)号：US20120170769A1
公开(公告)日：2012-07-05
申请号：US13265693
申请日：2010-04-29
申请人： David S. McGrath
发明人： David S. McGrath
IPC分类号： H03G5/00
CPC分类号： H03G9/18 , G10L21/0205 , G10L21/0264 , G10L21/0364 , H03G9/005 , H03G9/025
摘要： The loudness of an audio signal is adjusted while reducing changes in its perceived spectral balance, using a dynamically-controllable filter having a high-frequency response characteristic and a low-frequency response characteristic, controlled by dynamically-changing information on the desired gain in each of a plurality of frequency bands of the audio signal.
摘要翻译：使用具有高频响应特性和低频响应特性的动态可控滤波器来调节音频信号的响度，同时减少其感知频谱平衡的变化，该滤波器通过动态改变每个的音频信号的多个频带。

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