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1. 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。

2. 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。

3. 发明申请

US20130179175A1 Method and System for Encoding Audio Data with Adaptive Low Frequency Compensation 有权
标题翻译：用自适应低频补偿编码音频数据的方法和系统
公开(公告)号：US20130179175A1
公开(公告)日：2013-07-11
申请号：US13588890
申请日：2012-08-17
申请人： Arijit Biswas , Vinay Melkote , Michael Schug , Grant A. Davidson , Mark S. Vinton
发明人： Arijit Biswas , Vinay Melkote , Michael Schug , Grant A. Davidson , Mark S. Vinton
IPC分类号： G10L19/00
CPC分类号： G10L19/028 , G10L19/0204 , G10L19/032 , G10L19/265
摘要： A method for determining mantissa bit allocation of frequency domain audio data to be encoded, including by performing adaptive low frequency compensation on each frequency band of a set of low frequency bands of the data. The low frequency compensation includes steps of: performing tonality detection on the audio data to generate compensation control data indicative of whether each frequency band in the set has prominent tonal content; and performing low frequency compensation on each frequency band in the set having prominent tonal content, including by correcting a preliminary masking value for each frequency band having prominent tonal content, but not performing low frequency compensation on the audio data in any other frequency band in the set. Other aspects are audio encoding methods including such tonality detection and low frequency compensation steps, and a system configured to perform any embodiment of the inventive method.
摘要翻译：一种用于确定要编码的频域音频数据的尾数位分配的方法，包括通过对数据的一组低频带的每个频带执行自适应低频补偿。低频补偿包括以下步骤：对音频数据执行音调检测，以产生指示该组中的每个频带是否具有突出的音调内容的补偿控制数据; 并且对具有突出音调内容的集合中的每个频带执行低频补偿，包括通过校正具有突出音调内容的每个频带的初步屏蔽值，但不对低频补偿中的任何其他频带中的音频数据执行低频补偿组。其他方面是包括这种音调检测和低频补偿步骤的音频编码方法，以及被配置为执行本发明方法的任何实施例的系统。

4. 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。

5. 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。

6. 发明授权

US5394473A Adaptive-block-length, adaptive-transforn, and adaptive-window transform coder, decoder, and encoder/decoder for high-quality audio 失效
标题翻译：自适应块长度，自适应转换和自适应窗口变换编码器，解码器和编码器/解码器，用于高质量音频
公开(公告)号：US5394473A
公开(公告)日：1995-02-28
申请号：US781262
申请日：1991-12-11
申请人： Grant A. Davidson
发明人： Grant A. Davidson
IPC分类号： G05D11/13 , G10L19/02 , H03H17/00 , H03M7/30 , H04B1/66 , H04B14/00 , G10L5/00
CPC分类号： G10L19/0212 , H04B1/66 , G10L19/022
摘要： The invention relates in general to high-quality low bit-rate digital transform coding and decoding of information corresponding to audio signals such as music signals. More particularly, the invention relates to signal analysis/synthesis in coding and decoding. The invention can optimize the trade off in transform coders between time resolution and frequency resolution by adaptively selecting the transform block length for each sampled audio segment, and/or can optimize coding gain by adaptively selecting the transform and/or by adaptively selecting the analysis window or the analysis/synthesis window pair.
摘要翻译： PCT No.PCT / US91 / 02512 Sec。 371日期1991年12月11日 102（e）日期1991年12月11日PCT 1991年4月12日PCT PCT。 WO91 / 16769 PCT出版物日期1991年10月31日。本发明一般涉及对应于诸如音乐信号的音频信号的信息的高质量低比特率数字变换编码和解码。更具体地，本发明涉及编码和解码中的信号分析/合成。本发明可以通过自适应地选择每个采样音频段的变换块长度来优化时间分辨率和频率分辨率之间的变换编码器的折衷，和/或可以通过自适应地选择变换和/或通过自适应地选择分析窗口来优化编码增益或分析/综合窗口对。

7. 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。

8. 发明授权

US5451954A Quantization noise suppression for encoder/decoder system 失效
标题翻译：编码器/解码器系统的量化噪声抑制
公开(公告)号：US5451954A
公开(公告)日：1995-09-19
申请号：US101887
申请日：1993-08-04
申请人： Mark F. Davis , Craig C. Todd , Grant A. Davidson , Stephen D. Vernon
发明人： Mark F. Davis , Craig C. Todd , Grant A. Davidson , Stephen D. Vernon
IPC分类号： H04B1/66 , H04N5/21 , H04N7/26 , H04N7/30 , H03M7/00
CPC分类号： H04B1/665 , H04N19/126 , H04N19/124 , H04N19/60 , H04N5/21
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 the suppression of quantizing noise which results from low bit-rate encoding and decoding of signal events such as transients. An encoder identifies intervals of an input signal in which a decoder is likely to generate audible artifacts caused by quantizing errors. Indications of these intervals are passed in the encoded signal to the decoder. The decoder uses these indications to eliminate quantizing artifacts by suppressing corresponding intervals in the output signal.
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。更具体地说，本发明涉及由诸如瞬变的信号事件的低比特率编码和解码产生的量化噪声的抑制。编码器识别输入信号的间隔，其中解码器可能产生由量化误差引起的可听见的伪像。这些间隔的指示在编码信号中传递给解码器。解码器使用这些指示通过抑制输出信号中的相应间隔来消除量化伪像。

9. 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。

10. 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。

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