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    AUDIO PROCESSING USING AUDITORY SCENE ANALYSIS AND SPECTRAL SKEWNESS
    4.
    发明授权
    AUDIO PROCESSING USING AUDITORY SCENE ANALYSIS AND SPECTRAL SKEWNESS 有权
    声音处理听觉BY场景分析和频谱不对称

    公开(公告)号：EP2168122B1

    公开(公告)日：2011-11-30

    申请号：EP08780174.2

    申请日：2008-07-11

    申请人： Dolby Laboratories Licensing Corporation

    发明人： SEEFELDT, Alan, Jeffrey ,  SMITHERS, Michael, John

    IPC分类号： G10L21/02 ,  H03G3/20

    CPC分类号： G10L21/02 ,  H03G9/005 ,  H03G9/18

    摘要： A method for controlling the loudness of auditory events in an audio signal. In an embodiment, the method includes weighting the auditory events (an auditory event having a spectrum and a loudness), using skewness in the spectra and controlling loudness of the auditory events, using the weights. Various embodiments of the invention are as follows: The weighting being proportionate to the measure of skewness in the spectra; the measure of skewness is a measure of smoothed skewness; the weighting is insensitive to amplitude of the audio signal; the weighting is insensitive to power; the weighting is insensitive to loudness; and any relationship between signal measure and absolute reproduction level is not known at the time of weighting; the weighting includes weighting auditory-event-boundary importance, using skewness in the spectra.

    Automatic equalizing system with discrete Fourier transform (DFT) or fast Fourier transform (FFT)
    7.
    发明公开
    Automatic equalizing system with discrete Fourier transform (DFT) or fast Fourier transform (FFT) 失效
    变换使用离散傅立叶变换（DFT）自动均衡系统或快速傅立叶（FFT）。

    公开(公告)号：EP0119645A1

    公开(公告)日：1984-09-26

    申请号：EP84200221.4

    申请日：1984-02-20

    申请人： Philips Electronics N.V.

    发明人： Op de Beek, Franciscus J. ,  Van Nieuwland Jacob Maria

    IPC分类号： H04R3/04 ,  H04R29/00

    CPC分类号： H04R3/04 ,  H03G5/165 ,  H03G9/18 ,  H04S7/30

    摘要： The equalizing arrangement in accordance with the invention comprises an equalizing unit (4) for correcting the frequency characteristic of an electric signal applied to its first input (7) under the influence of a control signal (10) applied to its second input (8) and for supplying a corrected electric signal to its output (9), and a frequency analyzing unit (16) having a first input (15) coupled to a point in the signal path between the input terminal (1) and the first input (7) of the equalizing unit (4), a second input (17) coupled to the pick-up means (14) and an output (19) for supplying the control signal (10).
    The analyzing unit (16) is constructed to apply a Fourier transform to the previously digitized input signals x(t) and y(t) applied to its first and its second input, to compute Fourier-transformed signals F x (f) and Fy(f) of these two signals and to compute a transfer function between these two signals.
    Preferably the frequency analyzing unit is further constructed to compute a coherence function γ 2 zy (f) between the two signals and, in determining the control signal, to ignore the values of the transfer function for those frequencies f for which y 2 xy is smaller than a specific value, for example 0.8. By means of the arrangement in accordance with the invention a better equalization can be achieved and the distortion of the output signal of the arrangement is low.

    AUDIO SIGNAL AMPLITUDE SUPPRESSION DEVICE
    8.
    发明公开
    AUDIO SIGNAL AMPLITUDE SUPPRESSION DEVICE 审中-公开
    VORRICHTUNG ZURUNTERDRÜCKUNGEINER AUDIOSIGNALAMPLITUDE

    公开(公告)号：EP3010146A1

    公开(公告)日：2016-04-20

    申请号：EP14811383.0

    申请日：2014-03-26

    申请人： Large.Inc

    发明人： OSHIMA, Kenichi

    IPC分类号： H03G11/02 ,  H03G11/04 ,  H04R25/00

    CPC分类号： H04R25/356 ,  H03G9/18 ,  H03G11/02 ,  H03G11/04 ,  H04R1/1041 ,  H04R25/353 ,  H04R2225/43 ,  H04R2430/01 ,  H04R2430/03

    摘要： When, in conventional sound systems, signals of sounds collected by a microphone are amplified and the amplified signals are applied to an ear by an earphone and a headphone, too loud sounds and environmental noises are unpleasant, and the intelligibility of words are poor. Hence, such problems are to be addressed. In addition, downsizing and performance improvement for sound apparatuses are to be accomplished. A signal having undergone an amplitude limitation with reference to an output potential of a high cut filter 17 which is an output by a low cut filter 12 in a high frequency band and inverted by an inverter 14 is added with, by an adder 15, an output signal from the high cut filter 17, and an addition signal is output.

    摘要翻译： 当在传统的声音系统中，由麦克风收集的声音的信号被放大，放大的信号由耳机和耳机施加到耳朵时，声音太大，环境噪音不愉快，字的清晰度差。 因此，这些问题将得到解决。 此外，要完成声音设备的小型化和性能改进。 参考由高频带的低通滤波器12输出的反相器14反相的高频滤波器17的输出电位进行了幅度限制的信号，由加法器15加上 输出来自高频滤波器17的输出信号，并输出相加信号。

    A DIGITAL COMPRESSOR FOR COMPRESSING AN AUDIO SIGNAL
    9.
    发明公开
    A DIGITAL COMPRESSOR FOR COMPRESSING AN AUDIO SIGNAL 有权
    DIGITALER VERDICHTER ZUR KOMPRIMIERUNG EINES AUDIOSIGNALS

    公开(公告)号：EP2995000A1

    公开(公告)日：2016-03-16

    申请号：EP14702018.4

    申请日：2014-01-30

    申请人： Huawei Technologies Co., Ltd.

    发明人： GROSCHE, Peter ,  LANG, Yue ,  ZHANG, Qing

    IPC分类号： H03G7/00

    CPC分类号： H03G7/002 ,  H03G7/007 ,  H03G9/18

    摘要： A digital compressor for compressing an input audio signal is presented. The digital compressor comprises a compression gain control for providing a compression gain parameter, and a compression parameter determiner for determining a compression ratio from the compression gain parameter. The compression parameter determiner may be configured to weight the compression gain parameter by a predetermined weight factor to obtain the compression ratio. The digital compressor further comprises an auxiliary signal generator for manipulating the input audio signal in dependence of the compression ratio to obtain a first auxiliary signal, and a combiner unit for combining the first auxiliary signal with the compression gain parameter to obtain a second auxiliary signal, and for combining the input audio signal with the second auxiliary signal to obtain the compressed audio signal.

    摘要翻译： 提出了一种用于压缩输入音频信号的数字压缩器。 数字压缩器包括用于提供压缩增益参数的压缩增益控制和用于根据压缩增益参数确定压缩比的压缩参数确定器。 压缩参数确定器可以被配置为将压缩增益参数加权预定的权重因子以获得压缩比。 数字压缩器还包括辅助信号发生器，用于根据压缩比来操纵输入音频信号以获得第一辅助信号;以及组合器单元，用于将第一辅助信号与压缩增益参数组合以获得第二辅助信号， 并且用于将输入音频信号与第二辅助信号组合以获得压缩音频信号。

    METHOD AND SYSTEM FOR CONTROLLING DISTORTION IN A CRITICAL FREQUENCY BAND OF AN AUDIO SIGNAL
    10.
    发明公开
    METHOD AND SYSTEM FOR CONTROLLING DISTORTION IN A CRITICAL FREQUENCY BAND OF AN AUDIO SIGNAL 审中-公开
    方法和系统变形控制在一个关键的BAND的音频信号

    公开(公告)号：EP2606572A1

    公开(公告)日：2013-06-26

    申请号：EP11754565.7

    申请日：2011-08-11

    申请人： Dolby Laboratories Licensing Corporation

    发明人： POTARD, Guillaume

    IPC分类号： H03G9/18 ,  H04R3/04

    CPC分类号： H03G3/20 ,  H03G9/18 ,  H04S7/301

    摘要： In some embodiments, a method and system for controlling distortion of the output of a miniature speaker by attenuating critical frequency band of the input signal to be reproduced, using tuning parameters that have been predetermined where the critical frequency band is a frequency range of the speaker's frequency response in which Total Harmonic Distortion (THD) peaks. The distortion control is performed in a manner which allows an increase in the average loudness of the speaker's output without significantly increasing distortion. The tuning parameters include a center frequency and a bandwidth of the critical frequency band, and a power threshold value. In some embodiments, the system is a loudness maximizer configured to limit distortion of a speaker's output by limiting distortion in a critical frequency band using predetermined control parameters, and limit the dynamic range of the output signal and increase its perceived overall average loudness level.