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11. 发明授权

US5623577A Computationally efficient adaptive bit allocation for encoding method and apparatus with allowance for decoder spectral distortions 失效
标题翻译：用于解码器频谱失真的编码方法和装置的计算效率自适应比特分配
公开(公告)号：US5623577A
公开(公告)日：1997-04-22
申请号：US190655
申请日：1994-01-28
申请人： Louis D. Fielder
发明人： Louis D. Fielder
IPC分类号： G10L19/02 , H03M7/30 , H04B1/66 , H04B14/04 , G10L9/00
CPC分类号： H04B1/665 , H04B1/667
摘要： The invention relates in general to low bit-rate encoding and decoding of information such as audio information. More particularly, the invention relates to computationally efficient adaptive bit allocation and quantization of encoded information useful in high-quality low bit-rate coding systems.In audio applications, a digital split-band encoder splits an input signal into frequency subband signals having bandwidths commensurate with the critical bandwidths of the human auditory system, quantizes the subband signals according to values established by an allocation function, and assembles the quantized subband signals into an encoded signal. The allocation function establishes allocation values in accordance with psychoacoustic principles with allowance for decoding synthesis filter bank spectral distortions.In one embodiment, an allocation function establishes allocation values using a psychoacoustic masking threshold generated by estimating the power spectral density (PSD) of the input signal, generating an excitation pattern by applying a basilar-membrane spreading function to the PSD, adjusting the excitation pattern by an amount equal to a sensitivity function which specifies a signal-to-noise ratio (SNR) sufficient to achieve psychoacoustic masking, comparing the level of the adjusted pattern to the threshold of hearing and generating the psychoacoustic masking threshold which is equal to the larger of the two. An allocation function may allow for decoder synthesis filter bank spectral distortions in any of a number of ways such as by adapting the sensitivity function.
摘要翻译：本发明一般涉及诸如音频信息的信息的低比特率编码和解码。更具体地，本发明涉及在高质量低比特率编码系统中有用的编码信息的计算上有效的自适应比特分配和量化。在音频应用中，数字分离带编码器将输入信号分离成具有与人类听觉系统的关键带宽相称的带宽的频率子带信号，根据由分配函数建立的值量化子带信号，并且组合量化的子带信号变成编码信号。分配功能根据心理声学原理建立分配值，同时允许解码合成滤波器组频谱失真。在一个实施例中，分配功能使用通过估计输入信号的功率谱密度（PSD）产生的心理声学屏蔽阈值来建立分配值，通过向PSD施加基底膜扩展函数来产生激励模式，调整激励模式通过相当于灵敏度函数的量来确定足以实现心理声学掩蔽的信噪比（SNR），将调整后的图案的电平与听力阈值进行比较，并产生等于较大的心理声学屏蔽阈值的两个。分配功能可以允许以多种方式中的任一种的解码器合成滤波器组频谱失真，例如通过调整灵敏度函数。

12. 发明授权

US09462399B2 Audio playback system monitoring 有权
标题翻译：音频播放系统监控
公开(公告)号：US09462399B2
公开(公告)日：2016-10-04
申请号：US14126985
申请日：2012-06-27
申请人： Sunil Bharitkar , Brett G. Crockett , Louis D. Fielder , Michael Rockwell
发明人： Sunil Bharitkar , Brett G. Crockett , Louis D. Fielder , Michael Rockwell
IPC分类号： H04R29/00 , H04H60/33 , H04H60/04
CPC分类号： H04R29/002 , H04H60/04 , H04H60/33 , H04R3/12 , H04R29/001 , H04R2430/03
摘要： In some embodiments, a method for monitoring speakers within an audio playback system (e.g., movie theater) environment. In typical embodiments, the monitoring method assumes that initial characteristics of the speakers (e.g., a room response for each of the speakers) have been determined at an initial time, and relies on one or more microphones positioned in the environment to perform a status check on each of the speakers to identify whether a change to at least one characteristic of any of the speakers has occurred since the initial time. In other embodiments, the method processes data indicative of output of a microphone to monitor audience reaction to an audiovisual program. Other aspects include a system configured (e.g., programmed) to perform any embodiment of the inventive method, and a computer readable medium (e.g., a disc) which stores code for implementing any embodiment of the inventive method.
摘要翻译：在一些实施例中，一种用于监视音频播放系统（例如，电影院）环境中的扬声器的方法。在典型的实施例中，监视方法假设扬声器的初始特性（例如，每个扬声器的房间响应）已经在初始时间被确定，并且依赖于位于环境中的一个或多个麦克风来执行状态检查在每个扬声器上，以识别自从初始时间以来是否发生了任何扬声器的至少一个特征的改变。在其他实施例中，该方法处理指示麦克风的输出的数据，以监视对视听程序的观众反应。其他方面包括配置（例如，编程）以执行本发明方法的任何实施例的系统，以及存储用于实现本发明方法的任何实施例的代码的计算机可读介质（例如，盘）。

13. 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。

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