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公开(公告)号：US5509017A
公开(公告)日：1996-04-16
申请号：US232094
申请日：1994-04-29
摘要： A process for simultaneous transmission of signals from N signal sources via a corresponding number of transmission channels, in which the individual signals are divided into blocks and the blocks are transformed into spectral coefficients by transformation or filtering, the spectral coefficients undergoing a data reduction process. The blocks belonging to the individual signals are divided into sections. The respective current sections of all signals are processed simultaneously. The permissible interference for each section is determined utilizing a perception-specific model, and a request of currently required overall transmission capacity is calculated. The allotment of maximum transmission capacity at disposal for each individual signal is calculated from the overall transmission capacity at disposal and the currently required overall transmission capacity. Each signal is coded and transmitted with the thus determined capacity.
摘要翻译： PCT No.PCT / DE92 / 00905 Sec。 371日期1994年04月29日 102（e）日期1994年4月29日PCT提交1992年10月28日PCT公布。 出版物WO93 / 09645 日期：1993年5月13日。一种用于通过相应数量的传输信道从N个信号源同时传输信号的过程，其中各个信号被分成块，并且通过变换或滤波将块变换成频谱系数，频谱系数 正在进行数据简化过程。 属于各个信号的块被分成几个部分。 所有信号的各个电流部分同时被处理。 使用感知特定模型确定每个部分的容许干扰，并且计算当前所需的总传输容量的请求。 每个单独信号处理的最大传输容量的分配是根据处理的总体传输容量和当前所需的总传输容量计算的。 每个信号被编码并以这样确定的容量传输。
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公开(公告)号：US5742735A
公开(公告)日：1998-04-21
申请号：US295484
申请日：1994-08-25
申请人： Ernst Eberlein , Heinz Gerhauser , Harald Popp , Dieter Seitzer , Hartmut Schott , Karl Heinz Brandenburg
发明人： Ernst Eberlein , Heinz Gerhauser , Harald Popp , Dieter Seitzer , Hartmut Schott , Karl Heinz Brandenburg
CPC分类号： H04B1/667 , G10H1/0041 , G10H7/02 , G10L19/0208 , G10H2250/235
摘要： A digital adaptive transformation coding method for the transmission and/ortorage of audio signals, specifically music signals, wherein N scanned values of the audio signal are transformed into M spectral coefficients, and the coefficients are split up into frequency groups, quantized and then coded. The quantized maximum value of each frequency group is used to define the coarse variation of the spectrum. The same number of bits is assigned to all values in a frequency group. The bits are assigned to the individual frequency groups as a function of the quantized maximum value present in the particular frequency group. A multi-signal processor system is disclosed which is specifically designed for implementation of this method.
摘要翻译： 一种用于传输和/或存储音频信号，特别是音乐信号的数字自适应变换编码方法，其中音频信号的N个扫描值被变换为M个频谱系数，并将该系数分成频率组，然后被量化并随后编码 。 每个频率组的量化最大值用于定义频谱的粗略变化。 相同数量的位分配给频率组中的所有值。 这些位根据特定频率组中存在的量化最大值的函数分配给各个频率组。 公开了专门用于实施该方法的多信号处理器系统。
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公开(公告)号：US6101475A
公开(公告)日：2000-08-08
申请号：US696890
申请日：1996-08-21
申请人： Michael Keyhl , Harald Popp , Ernst Eberlein , Karl-Heinz Brandenburg , Heinz Gerhauser , Christian Schmidmer
发明人： Michael Keyhl , Harald Popp , Ernst Eberlein , Karl-Heinz Brandenburg , Heinz Gerhauser , Christian Schmidmer
CPC分类号： H04B1/665
摘要： In a method for the cascaded coding and decoding of audio data the spectral components of the short-time spectrum associated with a data block are formed for each data block with a certain number of time input data, the coded signal is formed, by quantization and coding, on the basis of the spectral components for this data block and using a psycho-acoustic model to determine the bit distribution for the spectral components, whereupon time output data are obtained by decoding at the end of each codec stage.To prevent a deterioration in the sound quality in codec cascades with a plurality of stages, an identification code is added to the coded signal at an initial stage to mark the start of the data block; furthermore, the subsequent codec stages divide the data blocks to be coded on the basis of this identification code.
摘要翻译： PCT No.PCT / EP94 / 03478 371日期：1996年8月21日 102（e）日期1996年8月21日PCT 1994年10月21日PCT PCT。 WO95 / 22858 PCT出版物 日期1995年8月24日在音频数据的级联编码和解码方法中，为每个具有一定数量时间输入数据的数据块形成与数据块相关联的短时频谱的频谱分量，编码信号为 通过量化和编码，基于该数据块的频谱分量形成，并使用心理声学模型来确定频谱分量的比特分布，由此在每个编解码器级结束时通过解码获得时间输出数据 。 为了防止多级编解码级联中的声音质量恶化，在初始阶段将识别码添加到编码信号，以标记数据块的开始; 此外，随后的编解码器级基于该识别码划分要编码的数据块。
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公开(公告)号：US5014318A
公开(公告)日：1991-05-07
申请号：US439394
申请日：1989-10-25
申请人： Hartmut Schott , Dieter Seitzer , Heinz Gerhauser , Karlheinz Brandenburg , Ernst Eberlein , Stefan Krageloh , Rolf Kapust , Harald Popp
发明人： Hartmut Schott , Dieter Seitzer , Heinz Gerhauser , Karlheinz Brandenburg , Ernst Eberlein , Stefan Krageloh , Rolf Kapust , Harald Popp
摘要： Disclosed is an apparatus for checking audio signal processing systems. The apparatus has the following features:the apparatus is provided with a first input connection, to which the input signal of the audio processing system to be checked is transmitted, a second input connection, to which the output signal of said system is transmitted, and a signal processor.said signal processor ascertains the signal delay time of said system to be checked by means of correlating said signals received at said two input connections,said signal processor always composes the difference signal from said signal received at said first input connection during a specific time span and said signal received at said second input connection, lagging by the signal delay time,said signal processor ascertains the spectral composition of said signal received at said first input connection during said specific time span and of said respective difference signal,said signal processor ascertains the hearing threshold of the human ear from said spectral composition and compares the ascertained hearing threshold with the respective difference signal.
摘要翻译： PCT No.PCT / DE89 / 00110 Sec。 371日期：1989年10月25日 102（e）日期1989年10月25日PCT提交1989年2月25日PCT公布。 公开号WO89 / 08357 日期为1989年9月8日。公开是用于检查音频信号处理系统的装置。 该装置具有以下特征：该装置具有第一输入连接，待检查的音频处理系统的输入信号被发送到该第一输入连接，传输所述系统的输出信号的第二输入连接;以及 信号处理器。 所述信号处理器通过将在所述两个输入连接处接收到的所述信号相关来确定要检查的所述系统的信号延迟时间，所述信号处理器总是在特定时间跨度期间组合来自在所述第一输入连接处接收的所述信号的差信号， 所述信号在所述第二输入连接处被接收，滞后于所述信号延迟时间，所述信号处理器确定在所述特定时间跨度期间在所述第一输入连接处接收到的所述信号的频谱组成以及所述各个差分信号，所述信号处理器确定听觉 阈值，并将所确定的听力阈值与相应的差分信号进行比较。
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公开(公告)号：US6115677A
公开(公告)日：2000-09-05
申请号：US43586
申请日：1998-03-20
申请人： Rainer Perthold , Heinz Gerhauser
发明人： Rainer Perthold , Heinz Gerhauser
CPC分类号： H01Q1/2233 , G08C17/02 , H01Q1/22 , H01Q9/0421
摘要： A consumption recording system affixed to a wall comprises a consumption ording device and a radio module, which is connected to the consumption recording device. A microstrip antenna is connected to the radio module. The microstrip antenna is placed within a non-metallic casing cover of the consumption recording device in such a way that the main radiation direction of the microstrip antenna is directed perpendicularly away from the wall.
摘要翻译： PCT No.PCT / EP96 / 04068 Sec。 371日期：1998年3月20日 102（e）1998年3月20日PCT PCT 1996年9月17日PCT公布。 公开号WO97 / 11445 PCT 日期1997年3月27日贴在墙上的消费记录系统包括消耗记录装置和连接到消费记录装置的无线电模块。 微带天线连接到无线电模块。 微带天线以这样的方式放置在消耗记录装置的非金属外壳盖内，使得微带天线的主辐射方向垂直于墙壁定向。
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公开(公告)号：USRE44897E1
公开(公告)日：2014-05-13
申请号：US13897221
申请日：2013-05-17
申请人： Oliver Kunz , Martin Dietz , Rainer Buchta , Jurgen Zeller , Karlheinz Brandenburg , Martin Sieler , Heinz Gerhauser
发明人： Oliver Kunz , Martin Dietz , Rainer Buchta , Jurgen Zeller , Karlheinz Brandenburg , Martin Sieler , Heinz Gerhauser
IPC分类号： H04B1/66
摘要： In a method for coding an audio signal digitized at a low sampling rate to obtain time domain audio samples. A frequency domain representation of the time domain audio samples is produced. The frequency domain representation includes successive frequency lines. These frequency lines are grouped into a plurality of scale factor bands. The successive frequency lines in a scale factor band are coded with the same scale factor. A plurality of regions is formed by grouping the scale factor bands, wherein successive scale factor bands form a region within which all the scale factors are coded with the same number of bits, which is determined according to the largest scale factor of the region. The scale factors assigned to scale factor bands within the highest region that includes the higher frequency successive frequency lines are set to zero. The frequency lines in the highest region are coded using the zero-valued scale factors that correspond to a multiplication factor of 1. The scale factors for the highest region, however, are not coded. Thus, the bits that would be required for coding these zero-valued scale factors are saved and can be used for a finer quantization of the rest of the spectrum. Additionally, this coding method when applied to ISO/IEC 13818-3 as a low sampling rate modification thereof only requires minimal changes with respect to this Standard.
摘要翻译： 在以低采样率对数字化的音频信号进行编码以获得时域音频样本的方法中， 产生时域音频样本的频域表示。 频域表示包括连续的频率线。 这些频率线被分组成多个比例因子频带。 缩放因子带中的连续频率线以相同的比例因子进行编码。 通过对比例因子频带进行分组来形成多个区域，其中，连续的比例因子波段形成一个区域，在该区域内，所有比例因子都以相同的比特数进行编码，这是根据该区域的最大比例因子确定的。 分配给包括较高频率的连续频率线的最高区域内的比例因子频带的比例因子被设置为零。 最高区域中的频率线使用与乘法因子1对应的零值比例因子进行编码。然而，最高区域的比例因子未被编码。 因此，编码这些零值比例因子所需的位被保存，并且可以用于其余频谱的更精细的量化。 此外，当将其应用于ISO / IEC 13818-3作为其低采样率修改时，该编码方法仅需要相对于本标准的最小变化。
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公开(公告)号：US06657492B1
公开(公告)日：2003-12-02
申请号：US09508654
申请日：2000-05-01
IPC分类号： H03F126
CPC分类号： H03F1/3247 , H03F2200/102
摘要： In a method for estimating a non-linear characteristic of an amplifier, an input signal is provided, a reference signal is determined based on the power of the input signal, the input signal is amplified using said amplifier in order to produce an amplified signal, a measurement signal is determined based on the power of the amplified signal and respective samples of the measurement signal are associated with respective samples of the reference signal. A gain value of said amplifier for each sample is determined from the respective samples of the reference signal and the measurement signal. An operating range of the amplifier is divided into a plurality of power sections and an averaged gain value is produced for each power section. Thereafter, a measure of the deviation of the gain values associated with a power section from said averaged gain value is determined. Then, averaged gain values are rejected which have a measure of the deviation which exceeds a predetermined threshold. Finally, an interpolation is performed using averaged gain values which have not been rejected in order to produce an estimation of said non-linear characteristic of said amplifier.
摘要翻译： 在用于估计放大器的非线性特性的方法中，提供输入信号，基于输入信号的功率确定参考信号，使用所述放大器放大输入信号以产生放大信号， 基于放大信号的功率确定测量信号，并且测量信号的相应采样与参考信号的相应采样相关联。 根据参考信号和测量信号的相应样本确定每个采样的所述放大器的增益值。 放大器的工作范围被分成多个功率部分，并且为每个功率部分产生平均增益值。 此后，确定与功率部分相关的增益值与所述平均增益值的偏差的度量。 然后，平均增益值被拒绝，其具有超过预定阈值的偏差量度。 最后，使用未被拒绝的平均增益值来执行内插，以便产生所述放大器的所述非线性特性的估计。
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公开(公告)号：US06424939B1
公开(公告)日：2002-07-23
申请号：US09402684
申请日：1999-10-06
申请人： Jürgen Herre , Uwe Gbur , Andreas Ehret , Martin Dietz , Bodo Teichmann , Oliver Kunz , Karlheinz Brandenburg , Heinz Gerhauser
发明人： Jürgen Herre , Uwe Gbur , Andreas Ehret , Martin Dietz , Bodo Teichmann , Oliver Kunz , Karlheinz Brandenburg , Heinz Gerhauser
IPC分类号： G10L1900
CPC分类号： H04B1/665 , G10L19/028
摘要： A method for coding or decoding an audio signal combines the advantages of TNS processing and noise substitution. A time-discrete audio signal is initially transformed to the frequency domain in order to obtain spectral values of the temporal audio signal. Subsequently, a prediction of the spectral values in relation to frequency is carried out in order to obtain spectral residual values. Within the spectral residual values, areas are detected encompassing spectral residual values with noise properties. The spectral residual values in the noise areas are noise-substituted, whereupon information concerning the noise areas and noise substitution is incorporated into side information pertaining to a coded audio signal. Thus, considerable bit savings in case of transient signals can be achieved.
摘要翻译： 用于对音频信号进行编码或解码的方法结合了TNS处理和噪声替换的优点。 时间离散音频信号最初被变换到频域以获得时间音频信号的频谱值。 随后，进行与频率相关的频谱值的预测，以获得谱残差值。 在光谱残差值内，检测到包含具有噪声特性的光谱残差值的区域。 噪声区域中的频谱残差值被噪声替代，因此关于噪声区域和噪声替换的信息被并入与编码音频信号有关的侧面信息中。 因此，可以实现在瞬态信号的情况下相当可观的位节省。
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公开(公告)号：US08798921B2
公开(公告)日：2014-08-05
申请号：US10362669
申请日：2001-08-14
IPC分类号： G01C21/00
CPC分类号： G08G1/0962 , G08G1/09675
摘要： A system for presenting information regarding an object provided within a plurality of information sources to a user or a presentation means depending on a location of the user or the presentation means includes a means for determining the location, a means for selecting the information to be presented depending on the determined location and on one or several pre-settable selection criteria defining an information source from the plurality of information sources, and a means for outputting the information to be presented.
摘要翻译： 根据用户或呈现装置的位置，向用户或呈现装置呈现关于在多个信息源内提供的对象的信息的系统包括用于确定位置的装置，用于选择要呈现的信息的装置 取决于确定的位置以及从多个信息源中定义信息源的一个或几个可预设的选择标准，以及用于输出要呈现的信息的装置。
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公开(公告)号：US06771777B1
公开(公告)日：2004-08-03
申请号：US09214656
申请日：1999-05-28
申请人： Uwe Gbur , Martin Dietz , Bodo Teichmann , Karlheinz Brandenburg , Heinz Gerhauser , Jürgen Herre , James Johnston
发明人： Uwe Gbur , Martin Dietz , Bodo Teichmann , Karlheinz Brandenburg , Heinz Gerhauser , Jürgen Herre , James Johnston
IPC分类号： H04H500
CPC分类号： H04S1/007
摘要： A method of coding stereo audio spectral values first carries out grouping of those values in scale factor bands, with which scale factors are associated. Sections are formed next, each comprising at least one scale factor band. The spectral values are coded within at least one section with a code book assigned to the section, out of a plurality of code books each with a code book number assigned to it, the number of the code book used being transmitted as side information to the coded stereo audio spectral values. At least one additional code book number is provided, which does not refer to a code book but shows information relevant to the section to which it is assigned. A method of decoding stereo audio spectral values which are partly coded by the intensity stereo process and which have side information uses the relevant information, showing the additional code book numbers, to cancel the existing coding of the stereo audio spectral values.
摘要翻译： 对立体声音频频谱值进行编码的方法首先对与比例因子相关联的比例因子频带中的那些值进行分组。 接下来形成切片，每个部分包括至少一个比例因子带。 频谱值在至少一个部分内被编码，其中分配有代码簿的部分，在分配有代码簿编号的多个代码簿中，使用的代码簿的编号作为辅助信息被发送到 编码立体声音频频谱值。 提供至少一个附加的代码簿编号，其不涉及代码簿，但是显示与其被分配的部分相关的信息。 解码由强度立体声处理部分地编码并且具有侧面信息的立体声音频频谱值的方法使用显示附加码本号码的相关信息来取消立体声音频频谱值的现有编码。
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