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1. 发明授权

US5142656A Low bit rate transform coder, decoder, and encoder/decoder for high-quality audio 失效
标题翻译：低比特率变换编码器，解码器和编码器/解码器，用于高品质音频
公开(公告)号：US5142656A
公开(公告)日：1992-08-25
申请号：US787541
申请日：1991-11-04
申请人： Louis D. Fielder , Grant A. Davidson
发明人： Louis D. Fielder , Grant A. Davidson
IPC分类号： G06T9/00 , G10L19/02 , H04B1/66
CPC分类号： G10L19/0212 , G06T9/005 , H04B1/665
摘要： A transform encoder, a transform decoder, and a transform encoder/decoder system employ adaptive bit allocation wherein each code word representing spectral information is allocated a fixed number of bits and an adaptive number of bits, except that at least some but not all code words are allocated a fixed number of bits.
摘要翻译：变换编码器，变换解码器和变换编码器/解码器系统采用自适应位分配，其中表示频谱信息的每个代码字被分配固定数量的位和自适应位数，除了至少一些但不是全部的代码字被分配固定位数。

2. 发明授权

US5109417A Low bit rate transform coder, decoder, and encoder/decoder for high-quality audio 失效
标题翻译：低比特率变换编码器，解码器和编码器/解码器，用于高品质音频
公开(公告)号：US5109417A
公开(公告)日：1992-04-28
申请号：US458894
申请日：1989-12-29
申请人： Louis D. Fielder , Grant A. Davidson
发明人： Louis D. Fielder , Grant A. Davidson
IPC分类号： G06T9/00 , G10L19/00 , G10L19/02 , H04B1/66
CPC分类号： H04B1/665 , G06T9/005 , G10L19/035
摘要： Quantizing noise is lessened in a speech signal system by using adaptive bit allocation for subband channels, wherein subband information of digital words is represented in block-floating-point form, and normalized mantiss as may allow dropping a sign bit.
摘要翻译：通过对子带通道使用自适应位分配，在语音信号系统中量化噪声减小，其中数字字的子带信息以块浮点形式表示，并且标准化的mantiss可以允许放弃符号位。

3. 发明授权

US5579404A Digital audio limiter 失效
标题翻译：数字音频限幅器
公开(公告)号：US5579404A
公开(公告)日：1996-11-26
申请号：US17535
申请日：1993-02-16
申请人： Louis D. Fielder , Marina Bosi-Goldberg , Grant A. Davidson , Kenneth J. Gundry
发明人： Louis D. Fielder , Marina Bosi-Goldberg , Grant A. Davidson , Kenneth J. Gundry
IPC分类号： H03G11/00 , H03G7/00 , H04H60/71
CPC分类号： H03G7/007 , H04H60/71
摘要： A signal processing system comprising components such as split-band perceptual coders can receive a peak amplitude limited input audio signal and can process the signal in such a manner that the processed signal preserves the apparent loudness of the input signal but is no longer peak-amplitude limited. In one embodiment, upsampling is used in estimating the resultant peak amplitude and gain factors established in response to the estimated peak amplitude are applied to one or more frequency subbands of the processed signal.
摘要翻译：包括分波段感知编码器等组件的信号处理系统可以接收峰值限幅输入音频信号，并且可以以这样的方式处理信号，使得处理的信号保持输入信号的明显响度，但不再是峰值幅度有限。在一个实施例中，上采样用于估计所得到的峰值幅度，并且响应于所估计的峰值幅度建立的增益因子被施加到处理信号的一个或多个频率子带。

4. 发明授权

US5632003A Computationally efficient adaptive bit allocation for coding method and apparatus 失效
标题翻译：用于编码方法和装置的计算有效的自适应比特分配
公开(公告)号：US5632003A
公开(公告)日：1997-05-20
申请号：US145975
申请日：1993-11-01
申请人： Grant A. Davidson , Craig C. Todd , Mark F. Davis , Brian D. Link , Louis D. Fielder
发明人： Grant A. Davidson , Craig C. Todd , Mark F. Davis , Brian D. Link , Louis D. Fielder
IPC分类号： H03M7/30 , H04B1/66 , H04B14/06 , 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 one embodiment, an audio split-band encoder splits an input signal into frequency subband signals, 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 based upon a masking threshold. The masking threshold is established by estimating the power spectral density (PSD) of the input signal, generating an excitation pattern by applying a spreading function to the PSD, adjusting the excitation pattern by an amount equal to a signal-to-noise ratio (SNR) offset sufficient to achieve psychoacoustic masking, comparing the level of the adjusted pattern to the threshold of hearing and generating a masking threshold which is equal to the larger of the two. The spreading function may be implemented by applying one or more IIR filters to the input signal PSD.
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。在一个实施例中，音频分离带编码器将输入信号分离成频率子带信号，根据由分配函数建立的值量化子带信号，并将量化的子带信号组合成编码信号。分配功能基于掩蔽阈值根据心理声学原理建立分配值。通过估计输入信号的功率谱密度（PSD）来建立屏蔽阈值，通过向PSD施加扩展函数来产生激励模式，将激励模式调整等于信噪比（SNR）的量）偏移足以实现心理声学掩蔽，将调整模式的电平与听力阈值进行比较，并产生等于两者中较大者的掩蔽阈值。可以通过将一个或多个IIR滤波器应用于输入信号PSD来实现扩展功能。

5. 发明授权

US09135907B2 Method and apparatus for reducing the effect of environmental noise on listeners 有权
标题翻译：降低环境噪音对听众的影响的方法和装置
公开(公告)号：US09135907B2
公开(公告)日：2015-09-15
申请号：US13703844
申请日：2011-06-16
申请人： Matthew C. Fellers , Alan J. Seefeldt , Brett G. Crockett , Grant A. Davidson , Louis D. Fielder
发明人： Matthew C. Fellers , Alan J. Seefeldt , Brett G. Crockett , Grant A. Davidson , Louis D. Fielder
IPC分类号： G10K11/16 , H03B29/00 , A61F11/06 , G10K11/00 , G10K11/178 , H04R1/10 , H04R3/00 , G10L21/0208
CPC分类号： G10K11/002 , G10K11/178 , G10K2210/1081 , G10L21/0208 , H04R1/1008 , H04R1/1083 , H04R3/005 , H04R2420/01
摘要： A method and apparatus for enhancing a desired audio signal for delivery through an electroacoustic channel include obtaining a noise estimate attributable to an external disturbance, applying the noise estimate to a dynamic noise compensation (DNC) process to thereby condition the desired audio signal as a function of the spectral characteristics of the noise estimate, applying the noise estimate to an adaptive equalization (AEQ) process to thereby condition the desired audio signal as a function of the electroacoustic response of the electroacoustic channel, and applying the noise estimate to an active noise cancellation (ANC) process configured to generate anti-noise for delivery into the electroacoustic channel.
摘要翻译：用于增强用于通过电声通道递送的期望音频信号的方法和装置包括获得归因于外部干扰的噪声估计，将噪声估计应用于动态噪声补偿（DNC）过程，从而将所需音频信号调节为函数的噪声估计的频谱特性，将噪声估计应用于自适应均衡（AEQ）过程，从而根据电声通道的电声响应来调节期望的音频信号，并将噪声估计应用于主动噪声消除（ANC）过程，其被配置为产生用于递送到电声通道中的抗噪声。

6. 发明申请

US20130083939A1 METHOD AND APPARATUS FOR REDUCING THE EFFECT OF ENVIRONMENTAL NOISE ON LISTENERS 有权
标题翻译：减少环境噪声对听力的影响的方法和装置
公开(公告)号：US20130083939A1
公开(公告)日：2013-04-04
申请号：US13703844
申请日：2011-06-16
申请人： Matthew C. Fellers , Alan J. Seefeldt , Brett G. Crockett , Grant A. Davidson , Louis D. Fielder
发明人： Matthew C. Fellers , Alan J. Seefeldt , Brett G. Crockett , Grant A. Davidson , Louis D. Fielder
IPC分类号： G10K11/00
CPC分类号： G10K11/002 , G10K11/178 , G10K2210/1081 , G10L21/0208 , H04R1/1008 , H04R1/1083 , H04R3/005 , H04R2420/01
摘要： A method and apparatus for enhancing a desired audio signal for delivery through an electroacoustic channel include obtaining a noise estimate attributable to an external disturbance, applying the noise estimate to a dynamic noise compensation (DNC) process to thereby condition the desired audio signal as a function of the spectral characteristics of the noise estimate, applying the noise estimate to an adaptive equalization (AEQ) process to thereby condition the desired audio signal as a function of the electroacoustic response of the electroacoustic channel, and applying the noise estimate to an active noise cancellation (ANC) process configured to generate anti-noise for delivery into the electroacoustic channel.
摘要翻译：用于增强用于通过电声通道递送的期望音频信号的方法和装置包括获得归因于外部干扰的噪声估计，将噪声估计应用于动态噪声补偿（DNC）过程，从而将所需音频信号调节为函数的噪声估计的频谱特性，将噪声估计应用于自适应均衡（AEQ）过程，从而根据电声通道的电声响应来调节期望的音频信号，并将噪声估计应用于主动噪声消除（ANC）过程，其被配置为产生用于递送到电声通道中的抗噪声。

7. 发明授权

US5479562A Method and apparatus for encoding and decoding audio information 失效
标题翻译：用于对音频信息进行编码和解码的方法和装置
公开(公告)号：US5479562A
公开(公告)日：1995-12-26
申请号：US79169
申请日：1993-06-18
申请人： Louis D. Fielder , Grant A. Davidson
发明人： Louis D. Fielder , Grant A. Davidson
IPC分类号： G06T9/00 , G10L19/14 , H04B1/66 , G10L9/00
CPC分类号： G06T9/005 , G10L19/0212 , G10L19/16 , H04B1/665
摘要： The invention relates to formatting encoded audio information in a form suitable for transmission or storage. Audio information is encoded into a binary form, using an invariant number of bits to represent at least some but not all of the encoded information. The information represented by an invariant number of bits is assembled into pre-established positions within a formatted frame.
摘要翻译：本发明涉及以适于传输或存储的形式格式化编码音频信息。音频信息被编码成二进制形式，使用不变数量的位来表示至少一些但不是全部的编码信息。由不变位数表示的信息被组合成格式化帧内的预先建立的位置。

8. 发明授权

US5230038A Low bit rate transform coder, decoder, and encoder/decoder for high-quality audio 失效
标题翻译：低比特率变换编码器，解码器和编码器/解码器，用于高品质音频
公开(公告)号：US5230038A
公开(公告)日：1993-07-20
申请号：US787665
申请日：1991-11-04
申请人： Louis D. Fielder , Grant A. Davidson
发明人： Louis D. Fielder , Grant A. Davidson
IPC分类号： G06T9/00 , G10L19/02 , G11B20/10 , H04B1/66
CPC分类号： G10L19/0212 , G06T9/005 , H04B1/665
摘要： A transform encoder, a transform decoder, and a transform encoder/decoder system utilize complex pre- and post-transform multiplication of input signal samples to implement concurrent application of a modified Discrete Cosine Transform and a modified Discrete Sine Transform according to the Evenly-Stacked Time Domain Aliasing Cancellation technique against two channels of input signal samples, and to reduce the computational complexity of a digital filter bank of a modified Discrete Cosine Transform.
摘要翻译：变换编码器，变换解码器和变换编码器/解码器系统利用输入信号采样的复杂的变换前和变换后乘法来实现修正离散余弦变换和经均匀叠加的修正离散正弦变换的并行应用针对两个通道的输入信号样本的时域混叠消除技术，并减少了改进的离散余弦变换的数字滤波器组的计算复杂度。

9. 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。

10. 发明授权

US5357594A Encoding and decoding using specially designed pairs of analysis and synthesis windows 失效
标题翻译：使用专门设计的分析和综合窗口对进行编码和解码
公开(公告)号：US5357594A
公开(公告)日：1994-10-18
申请号：US78594
申请日：1993-06-16
申请人： Louis D. Fielder
发明人： Louis D. Fielder
IPC分类号： G06T9/00 , G10L19/02 , G11B20/10 , H04B1/66 , G10L3/02
CPC分类号： G10L19/0212 , G06T9/005 , H04B1/665
摘要： The invention relates to the design of analysis and synthesis windows for use in high-quality transform encoding and decoding of audio signals, especially encoding and decoding having a short signal-propagation delay. The design method derives a pair of analysis/synthesis windows from a known window function which satisfy various filter selectivity and window overlap-add constraints.
摘要翻译：本发明涉及用于音频信号的高质量变换编码和解码的分析和合成窗口的设计，特别是具有短信号传播延迟的编码和解码。设计方法从已知的窗口函数中得到一对分析/综合窗口，该窗口函数满足各种过滤器选择性和窗口重叠加法约束。

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