公开(公告)号：US08976979B2

公开(公告)日：2015-03-10

申请号：US13300480

申请日：2011-11-18

Applicant: Brett G. Crockett ,  Alan J. Seefeldt ,  Brian D. Link ,  John D. Couling

Inventor： Brett G. Crockett ,  Alan J. Seefeldt ,  Brian D. Link ,  John D. Couling

IPC: H03G5/00 ,  G10L21/02 ,  G11B20/10 ,  H03G5/16 ,  H04B1/00

CPC classification number: G10L21/0205 ,  G10L21/0364 ,  G11B20/10527 ,  G11B2020/10546 ,  G11B2020/10555 ,  G11B2020/10564 ,  H03G5/165

Abstract: Apparatuses for and methods of carrying out dynamic equalization processing of an audio signal, and apparatuses for and methods of controlling such equalization processing of the audio signal to dynamically adjust the time-varying spectrum of an audio signal to more closely match a user specified target time-invariant perceived audio signal spectrum while preserving the original dynamic range of the audio signal. The dynamic equalization is according to a user-defined spectral profile specified by a control interface that allows a user to define, create, modify and/or apply the user-defined spectral profile.

Abstract translation: 用于执行音频信号的动态均衡处理的装置和方法，以及用于控制音频信号的这种均衡处理的装置和方法，以动态地调整音频信号的时变频谱以更紧密地匹配用户指定的目标时间 - 不变的感知音频信号频谱，同时保持音频信号的原始动态范围。 动态均衡是根据由用户定义的，由用户定义，修改和/或应用用户定义的频谱分布的控制接口指定的用户定义的频谱分布。

公开(公告)号：US20120063614A1

公开(公告)日：2012-03-15

申请号：US13300480

申请日：2011-11-18

Applicant: Brett G. Crockett ,  Alan J. Seefeldt ,  Brian D. Link ,  John D. Couling

Inventor： Brett G. Crockett ,  Alan J. Seefeldt ,  Brian D. Link ,  John D. Couling

IPC: H03G5/00

CPC classification number: G10L21/0205 ,  G10L21/0364 ,  G11B20/10527 ,  G11B2020/10546 ,  G11B2020/10555 ,  G11B2020/10564 ,  H03G5/165

Abstract: Apparatuses for and methods of carrying out dynamic equalization processing of an audio signal, and apparatuses for and methods of controlling such equalization processing of the audio signal to dynamically adjust the time-varying spectrum of an audio signal to more closely match a user specified target time-invariant perceived audio signal spectrum while preserving the original dynamic range of the audio signal. The dynamic equalization is according to a user-defined spectral profile specified by a control interface that allows a user to define, create, modify and/or apply the user-defined spectral profile.

Abstract translation: 用于执行音频信号的动态均衡处理的装置和方法，以及用于控制音频信号的这种均衡处理的装置和方法，以动态地调整音频信号的时变频谱以更紧密地匹配用户指定的目标时间 - 不变的感知音频信号频谱，同时保持音频信号的原始动态范围。 动态均衡是根据由用户定义的，由用户定义，修改和/或应用用户定义的频谱分布的控制接口指定的用户定义的频谱分布。

公开(公告)号：US5632003A

公开(公告)日：1997-05-20

申请号：US145975

申请日：1993-11-01

Applicant: Grant A. Davidson ,  Craig C. Todd ,  Mark F. Davis ,  Brian D. Link ,  Louis D. Fielder

Inventor： Grant A. Davidson ,  Craig C. Todd ,  Mark F. Davis ,  Brian D. Link ,  Louis D. Fielder

IPC: H03M7/30 ,  H04B1/66 ,  H04B14/06 ,  G10L9/00

CPC classification number: H04B1/665 ,  H04B1/667

Abstract: 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.

Abstract translation: 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。