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

IPC分类号： G10H1/00 ,  G10H7/02 ,  G10L19/02 ,  H04B1/66 ,  G10L9/00

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。

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

IPC分类号： G10L11/00 ,  G10L21/02 ,  G11B20/18 ,  H04B14/04 ,  H04H20/88

CPC分类号： H04H20/88 ,  H04B14/04

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。

公开(公告)号：US06584138B1

公开(公告)日：2003-06-24

申请号：US09142325

申请日：1998-10-05

申请人： Christian Neubauer ,  Ernst Eberlein ,  Roland Plankenbühler ,  Albert Heuberger ,  Heinz Gerhäuser ,  Rainer Perthold ,  Hartmut Schott

发明人： Christian Neubauer ,  Ernst Eberlein ,  Roland Plankenbühler ,  Albert Heuberger ,  Heinz Gerhäuser ,  Rainer Perthold ,  Hartmut Schott

IPC分类号： H04B1500

CPC分类号： H04H20/31

摘要： In a coding method and a coder for introducing a non-audible data signal into an audio signal, the audio signal is first transformed to a spectral range and the masking threshold of the audio signal is determined. A pseudo-noise signal and a data signal are provided and multiplied with each other so a to provide a frequency-spread data signal. The spread data signal is weighted with the masking threshold, and thereafter the audio signal and the weighted data signal are superimposed. In a method and a decoder for decoding a data signal introduced into an audio signal in non-audible manner, the audio signal is first sampled and thereafter the sampled audio signal is filtered in non-recursive manner. The filtered audio signal is subsequently compared to a threshold value so as to retrieve the data signal.

摘要翻译： 在用于将非可听数据信号引入音频信号的编码方法和编码器中，首先将音频信号变换为频谱范围，并确定音频信号的掩蔽阈值。 提供伪噪声信号和数据信号并将其相乘，以便提供频率扩展的数据信号。 扩展数据信号用掩蔽阈值加权，之后叠加音频信号和加权数据信号。 在用于以不可听的方式对引入到音频信号中的数据信号进行解码的方法和解码器中，首先对音频信号进行采样，然后以非递归的方式对采样的音频信号进行滤波。 滤波后的音频信号随后与阈值进行比较，以检索数据信号。