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公开(公告)号：EP3100353A1
公开(公告)日：2016-12-07
申请号：EP14702793.2
申请日：2014-01-30
Applicant: Huawei Technologies Co., Ltd.
Inventor： GROSCHE, Peter , LANG, Yue , ZHANG, Qing
IPC: H03G7/00
Abstract: The invention relates to an audio compression system (100) for compressing an input audio signal, the audio compression system (100) comprising a digital filter (101) for filtering the input audio signal, the digital filter (101) comprising a frequency transfer function having a magnitude over frequency, the magnitude being formed by an equal loudness curve of a human ear to obtain a filtered audio signal, and a compressor (103) being configured to compress the input audio signal upon the basis of the filtered audio signal to obtain a compressed audio signal.
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2.发明公开CONCEPT FOR COMBINED DYNAMIC RANGE COMPRESSION AND GUIDED CLIPPING PREVENTION FOR AUDIO DEVICES 审中-公开CONCEPT在合并后的动态范围压缩，转移削波预防音频设备
公开(公告)号：EP3061090A1
公开(公告)日：2016-08-31
申请号：EP14786881.4
申请日：2014-10-20
Inventor： KÜCH, Fabian , UHLE, Christian , KRATSCHMER, Michael , NEUGEBAUER, Bernhard , MEIER, Michael , BORSUM, Arne
IPC: G10L21/034 , H03G9/00 , H03G11/00 , G10L19/16 , G10L19/008
CPC classification number: G10L19/167 , G06F3/165 , G10L19/008 , G10L19/018 , G10L21/034 , H03G9/005 , H03G11/008
Abstract: The invention provides a concept for combined dynamic range compression and guided clipping prevention for audio devices. An audio decoder for decoding an audio bitstream and a metadata bitstream related to the audio bitstream according to the concept includes an audio processing chain including a plurality of adjustment stages including a dynamic range control stage for adjusting a dynamic range of the audio output signal and a guided clipping prevention stage for preventing clipping of the audio output signal; and a metadata decoder configured to receive the metadata bitstream and to extract dynamic range control gain sequences and guided clipping prevention gain sequences from the metadata bitstream, at least a part of the dynamic range control gain sequences being supplied to the dynamic range control stage, and at least a part of the guided clipping prevention gain sequences being supplied to the guided clipping prevention stage.
Abstract translation: 本发明提供了组合的动态范围压缩为音频设备的一个概念和引导削波防止。 对于音频比特流的解码和与音频位流雅丁到概念的元数据的比特流的音频解码器包括在音频处理链包括调节阶段，包括用于调整所述音频输出信号和动态范围的动态范围控制级一个多元化 引导剪裁预防阶段防止音频输出信号的限幅; 和元数据解码器配置为接收所述元数据的比特流，并从元数据的比特流中提取的动态范围控制增益序列和引导削波防止增益序列，至少被提供给动态范围控制级的动态范围控制增益序列的一部分，和 至少被引导削波防止增益序列的一部分被供应到所述引导裁剪预防阶段。
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公开(公告)号：EP2312747B1
公开(公告)日：2013-09-04
申请号：EP10189702.3
申请日：2006-12-06
Inventor： Furge, Kenneth, Carl
CPC classification number: H03G7/004 , H03F1/32 , H03F2200/03 , H03F2200/321 , H03F2200/471 , H03F2200/66 , H03F2200/78 , H03G3/007 , H03G11/008
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公开(公告)号：EP1192557A4
公开(公告)日：2005-03-23
申请号：EP00923462
申请日：2000-04-14
Applicant: ORBAN INC A HARMAN INTERNAT CO
Inventor： ORBAN ROBERT A
CPC classification number: H03G11/008
Abstract: A method for computing the "clippings" of an audio signal in the digital domain to prevent aliasing is disclosed. An offset signal is found (10) by subtracting a threshold from samples of the audio signal (12). These are multiplied by a pulse (14) after the pulse is lined up in time with the crossing of the audio signal at the threshold (16). The pulse is the sum of two step functions (18), both of which are bandwidth-limited.
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公开(公告)号：EP1142109A4
公开(公告)日：2004-07-14
申请号：EP99966371
申请日：1999-12-17
Applicant: ORBAN INC A HARMAN INTERNAT CO
Inventor： ORBAN ROBERT A
CPC classification number: H03G11/008
Abstract: A digital clipper is highly oversampled to decrease aliasing and increase accuracy. The difference between the clipper's input (50) and output (80) is then downsampled and added to the delayed, unclipped signal at Ix sample rate to achieve clipping. Filters (90) operating at Ix can be placed in series with the downsampled differentially-clipped signal to achieve overshoot compensation, bandlimiting of the clipped signal, and other goals.
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Patent Agency Ranking
公开(公告)号：EP1067683A1
公开(公告)日：2001-01-10
申请号：EP99440110.7
申请日：1999-05-14
Applicant: ALCATEL
Inventor： Mortensen, Ivar, Dr.-Ing. , Neustadt, Alf, Dipl.-Ing.
CPC classification number: H04B1/707 , H03G7/007 , H03G11/008 , H04B2201/70706
Abstract: A method is described of generating an electrical output signal (CFCS) from an electrical input signal (CS) wherein said input signal (CS) is first clipped (12) and then filtered (13) into said output signal (CFCS). The invention is characterized in that said input signal (CS) is clipped dependant on said succeeding filtering. Thereby, a finite power peak capacity of a succeeding amplifier is not exceeded.
Abstract translation: 描述了一种从电输入信号（CS）生成电输出信号（CFCS）的方法，其中所述输入信号（CS）首先被剪切（12），然后被滤波（13）到所述输出信号（CFCS）中。 本发明的特征在于所述输入信号（CS）取决于所述后续滤波。 从而，不会超过后续放大器的有限功率峰值容量。
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公开(公告)号：EP0173452A3
公开(公告)日：1986-08-27
申请号：EP85305281
申请日：1985-07-24
Applicant: FORD AEROSPACE & COMMUNICATIONS CORPORATION
Inventor： Wisniewski, Jacob H.
IPC: H03G11/00
CPC classification number: H03G11/008
Abstract: A soft digital limiter (2) for limiting an analog input signal (1) from a maximum expected range (61) to a useful range (60). The number (m) of desired levels of resolution in the limiter (2) is preselected to be any power of two. An analog-to-digital converter (9) converts the input analog signal (1) to a digital representation (20). The converter (9) has its input voltage rating matched to the maximum expected range (61) and its output resolution matched to the preselected degree (m) of resolution. In the preferred two's complement numbering system, the condition for the input signal (1) falling within the useful range (60) is that the most significant p + 1 bits of the digital representation (20) are all identical, where p is the number of bits required by the converter (9) to delineate that portion of the maximum expected range (61) outside of the useful range (60). A network of comparators (e.g., 38. 39) implements this condition.
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公开(公告)号：EP2595312B1
公开(公告)日：2018-02-28
申请号：EP11189633.8
申请日：2011-11-17
Applicant: Dialog Semiconductor B.V.
Inventor： Helsloot, Michiel, Andre
CPC classification number: H03F3/181 , H03F1/3223 , H03F1/34 , H03F2200/03 , H03F2200/105 , H03G3/04 , H03G3/301 , H03G3/3089 , H03G11/00 , H03G11/008 , H04R3/007 , H04R3/02 , H04R3/04 , H04R29/003 , H04R2499/11
Abstract: A method of attenuating an input signal to obtain an output signal is described. The method comprises receiving the input signal, attenuating the input signal with a gain factor to obtain the output signal, applying a filter having a frequency response with a frequency-dependent filter gain to at least one of a copy of the input signal and a copy of the output signal to obtain a filtered signal, the frequency-dependent filter gain being arranged to emphasize frequencies within a number N of predetermined frequency ranges, N>1; wherein the filter comprises a sequence ofN sub-filters, each one of the N sub-filters having a frequency response adapted to emphasize frequencies within a corresponding one of the N predetermined frequency ranges; determining a signal strength of the filtered signal, and determining the gain factor from at least the signal strength.
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公开(公告)号：EP3089161A1
公开(公告)日：2016-11-02
申请号：EP14873206.8
申请日：2014-12-12
Applicant: Sony Corporation
Inventor： YAMAMOTO Yuki , CHINEN Toru , HONMA Hiroyuki , SHI Runyu
IPC: G10L19/00
CPC classification number: G10L19/02 , G10L19/26 , G10L21/034 , H03G7/002 , H03G7/007 , H03G9/00 , H03G9/005 , H03G11/008
Abstract: The present technology relates to a decoding apparatus, a decoding method and a program which make it possible to obtain sound with higher quality.
A demultiplexing circuit demultiplexes an input code string into a gain code string and a signal code string. A signal decoding circuit decodes the signal code string to output a time series signal. A gain decoding circuit decodes the gain code string. That is, the gain decoding circuit reads out gain values and gain inclination values at predetermined gain sample positions of the time series signal and interpolation mode information. An interpolation processing unit obtains a gain value at each sample position between two gain sample positions through linear interpolation or non-linear interpolation according to the interpolation mode based on the gain values and the gain inclination values. A gain applying circuit adjusts a gain of the time series signal based on the gain values. The present technology can be applied to a decoding apparatus.Abstract translation: 本技术涉及使得可以获得更高质量的声音的解码装置，解码方法和程序。 解复用电路将输入码串解复用为增益码串和信号码串。 信号解码电路解码信号码串以输出时间序列信号。 增益解码电路解码增益代码串。 也就是说，增益解码电路在时间序列信号和插值模式信息的预定增益采样位置读出增益值和增益倾斜值。 内插处理单元根据基于增益值和增益倾斜值的内插模式，通过线性内插或非线性内插，在两个增益采样位置之间的各取样位置获取增益值。 增益应用电路根据增益值调整时间序列信号的增益。 本技术可以应用于解码装置。
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公开(公告)号：EP1798852B1
公开(公告)日：2014-11-05
申请号：EP06025257.4
申请日：2006-12-06
Inventor： Furge, Kenneth Carl
CPC classification number: H03G7/004 , H03F1/32 , H03F2200/03 , H03F2200/321 , H03F2200/471 , H03F2200/66 , H03F2200/78 , H03G3/007 , H03G11/008
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