专利快速检索-快速检索全球专利，免费商用专利数据库-IPRDB

1. 发明授权

US06341165B1 Coding and decoding of audio signals by using intensity stereo and prediction processes 有权
标题翻译：通过使用强度立体声和预测过程对音频信号进行编码和解码
公开(公告)号：US06341165B1
公开(公告)日：2002-01-22
申请号：US09214655
申请日：1999-05-28
申请人： Uwe Gbur , Martin Dietz , Karlheinz Brandenburg , Heinz Gerhauser , Jürgen Herre , Schuyler Quackenbush
发明人： Uwe Gbur , Martin Dietz , Karlheinz Brandenburg , Heinz Gerhauser , Jürgen Herre , Schuyler Quackenbush
IPC分类号： H04R500
CPC分类号： H04B1/665 , G10L19/008 , G10L19/04 , H04B1/667 , H04B14/068 , H04S1/007 , H04S3/00
摘要： In the coding and decoding of stereo audio spectral values both the intensity stereo process and prediction are used in order to achieve high data compression. If intensity stereo coding is active in one section of scale factor bands, the prediction for the right channel in that range is deactivated, whereby the results of the prediction are not used to form the coded stereo audio spectral values. To allow further adaptation of the prediction for the right channel, the predictor of the right channel is fed with stereo audio spectral values for the channel, which again are intensity stereo decoded.
摘要翻译：在立体声音频频谱值的编码和解码中，使用强度立体声处理和预测来实现高数据压缩。如果强度立体声编码在比例因子频带的一个部分中有效，则该范围内的右声道的预测被去激活，由此预测结果不用于形成编码的立体声音频频谱值。为了允许对右声道的预测的进一步调整，右声道的预测器被馈送用于声道的立体声音频频谱值，其再次是强度立体声解码。

2. 发明授权

US06424939B1 Method for coding an audio signal 有权
标题翻译：音频信号编码方法
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。

3. 发明授权

US06771777B1 Process for coding and decoding stereophonic spectral values 有权
标题翻译：立体声频谱值的编码和解码过程
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。

4. 发明授权

US06766293B1 Method for signalling a noise substitution during audio signal coding 有权
标题翻译：在音频信号编码期间用信号通知噪声替换的方法
公开(公告)号：US06766293B1
公开(公告)日：2004-07-20
申请号：US09367775
申请日：1999-08-18
申请人： Jürgen Herre , Uwe Gbur , Andreas Ehret , Martin Dietz , Bodo Teichmann , Oliver Kunz , Karlheinz Brandenburg , Heinz Gerhäuser
发明人： Jürgen Herre , Uwe Gbur , Andreas Ehret , Martin Dietz , Bodo Teichmann , Oliver Kunz , Karlheinz Brandenburg , Heinz Gerhäuser
IPC分类号： G10L2102
CPC分类号： G10L19/028 , H04B1/665
摘要： In a method for signalling a noise substitution when coding an audio signal, the time-domain audio signal is first transformed into the frequency domain to obtain spectral values. The spectral values are subsequently grouped together to form groups of spectral values. On the basis of a detection establishing whether a group of spectral values is a noisy group or not, a codebook is allocated to a non-noisy or tonal group by means of a codebook number for redundancy coding of the same. If a group is noisy, an additional codebook number which does not refer to a codebook is allocated to it in order to signal that this group is noisy and therefore does not have to be redundancy coded. By signalling noise substitution by means of a Huffman codebook number for noisy groups of spectral values, which are e.g. sections made up of scale factor bands which do not have to be redundancy coded, an opportunity is provided to indicate the presence of a noise substitution in a scale factor band in the bit stream syntax of the MPEG-2 Advanced Audio Coding (AAC) Standard without having to interfere with the basic coding structure and without having to meddle with the structure of the existing bit stream syntax.
摘要翻译：在对音频信号编码时用于发信号通知的方法中，首先将时域音频信号变换成频域以获得频谱值。光谱值随后被分组在一起以形成光谱值组。基于确定一组频谱值是否为噪声组的检测，通过用于冗余编码的码本号将码本分配给非噪声或色调组。如果组噪声，则分配不附加码本的附加码本号，以便发信号通知该组噪声，因此不必进行冗余编码。通过用于噪声组的频谱值的霍夫曼码本号对信号进行信号替换，由不必冗余编码的比例因子带组成的部分提供了一种机会，以指示在MPEG-2高级音频编码（AAC）标准的比特流语法中的比例因子频带中存在噪声替换而不必干扰基本编码结构，而不必介入现有比特流语法的结构。

5. 发明授权

US06502069B1 Method and a device for coding audio signals and a method and a device for decoding a bit stream 有权
标题翻译：用于编码音频信号的方法和装置以及用于解码比特流的方法和装置
公开(公告)号：US06502069B1
公开(公告)日：2002-12-31
申请号：US09530001
申请日：2000-04-20
申请人： Bernhard Grill , Jürgen Herre , Bodo Teichmann , Karlheinz Brandenburg , Heinz Gerhauser
发明人： Bernhard Grill , Jürgen Herre , Bodo Teichmann , Karlheinz Brandenburg , Heinz Gerhauser
IPC分类号： G10L1912
CPC分类号： H04B1/665 , H04B14/046
摘要： The present invention permits a combination of a scalable audio coder with the TNS technique. In a method for coding time signals sampled in a first sampling rate, second time signals are first generated whose sampling rate is smaller than the first sampling rate. The second time signals are then coded according to a first coding algorithm and written into a bit stream. The coded second time signals are, however, decoded again, and, like the first time signals, transformed into the frequency domain. From a spectral representation of the first time signals, TNS prediction coefficients are calculated. The transformed output signal of the coder/decoder with the first coding algorithm, like the spectral representation of the first time signal, undergoes a prediction over the frequency to obtain residual spectral values for both signals, though only the prediction coefficients calculated on the basis of the first time signals are used. These two signals are evaluated against each other. The evaluated residual spectral values are then coded by means of a second coding algorithm to obtain coded evaluated residual spectral values, which, together with the side information containing the calculated prediction coefficients, are written into the bit stream.
摘要翻译：本发明允许可扩展音频编码器与TNS技术的组合。在对以第一采样率采样的时间信号进行编码的方法中，首先生成采样率小于第一采样率的第二时间信号。然后根据第一编码算法对第二时间信号进行编码并写入比特流。然而，编码的第二时间信号被再次解码，并且像第一次信号一样被转换成频域。根据第一时间信号的频谱表示，计算TNS预测系数。使用第一编码算法的编码器/解码器的变换输出信号，如第一时间信号的频谱表示，对频率进行预测，以获得两个信号的残差频谱值，尽管仅基于第一次使用信号。这两个信号被相互评估。然后通过第二编码算法对所评估的残差频谱值进行编码，以获得编码的估计残差频谱值，其与包含计算的预测系数的边信息一起写入比特流。

6. 再颁专利

USRE44897E1 Process of low sampling rate digital encoding of audio signals 有权
标题翻译：音频信号低采样率数字编码过程
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。

7. 发明授权

US5848391A Method subband of coding and decoding audio signals using variable length windows 失效
标题翻译：使用可变长度窗口对音频信号进行编码和解码的方法子带
公开(公告)号：US5848391A
公开(公告)日：1998-12-08
申请号：US678666
申请日：1996-07-11
申请人： Marina Bosi , Grant Davidson , Charles Robinson , Martin Dietz , Uwe Gbur , Oliver Kunz , Karlheinz Brandenburg
发明人： Marina Bosi , Grant Davidson , Charles Robinson , Martin Dietz , Uwe Gbur , Oliver Kunz , Karlheinz Brandenburg
IPC分类号： G10L19/02 , H04B1/66 , G10L5/00
CPC分类号： G10L19/022 , H04B1/665
摘要： A method of encoding time-discrete audio signals comprises the steps of weighting the time-discrete audio signal by means of window functions overlapping each other so as to form blocks, the window functions producing blocks of a first length for signals varying weakly with time and blocks of a second length for signals varying strongly with time. A start window sequence is selected for the transition from windowing with blocks of the first length to windowing with blocks of the second length, whereas a stop window sequence is selected for the opposite transition. The start window sequence is selected from at least two different start window sequences having different lengths, whereas the stop window sequence is selected from at least two different stop window sequences having different lengths. A method of decoding blocks of encoded audio signals selects a suitable inverse transformation as well as a suitable synthesis window as a reaction to side information associated with each block.
摘要翻译：对时间离散音频信号进行编码的方法包括以下步骤：通过彼此重叠的窗口函数对时间离散音频信号进行加权，以形成块，窗口函数产生用于随时间变化的信号的第一长度的块，对于随时间强烈变化的信号的第二长度的块。选择起始窗口序列，用于从具有第一长度的块的窗口到具有第二长度的块的窗口的转换，而针对相反的转换选择停止窗口序列。起始窗口序列从具有不同长度的至少两个不同的开始窗口序列中选择，而停止窗口序列从具有不同长度的至少两个不同的停止窗口序列中选择。对编码音频信号的块进行解码的方法选择合适的逆变换以及合适的合成窗口作为与每个块相关联的侧信息的反应。

8. 发明授权

US5014318A Apparatus for checking audio signal processing systems 失效
标题翻译：用于检查音频信号处理系统的装置
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。

9. 发明授权

US07512539B2 Method and device for processing time-discrete audio sampled values 有权
标题翻译：用于处理时间离散音频采样值的方法和装置
公开(公告)号：US07512539B2
公开(公告)日：2009-03-31
申请号：US10479398
申请日：2002-05-28
申请人： Ralf Geiger , Thomas Sporer , Karlheinz Brandenburg , Jürgen Herre , Jürgen Koller
发明人： Ralf Geiger , Thomas Sporer , Karlheinz Brandenburg , Jürgen Herre , Jürgen Koller
IPC分类号： G06F17/14 , G10L19/00
CPC分类号： G10L19/0212 , G06F17/147
摘要： An integer transform, which provides integer output values, carries out the TDAC function of a MDCT in the time domain before the forward transform. In overlapping windows, this results in a Givens rotation which may be represented by lifting matrices, wherein time-discrete sampled values of an audio signal may at first be summed up on a pair-wise basis to build a vector so as to be sequentially provided with a lifting matrix. After each multiplication of a vector by a lifting matrix, a rounding step is carried out such that, on the output-side, only integers will result. By transforming the windowed integer sampled value with an integer transform, a spectral representation with integer spectral values may be obtained. The inverse mapping with an inverse rotation matrix and corresponding inverse lifting matrices results in an exact reconstruction.
摘要翻译：提供整数输出值的整数变换在正向变换之前的时域中执行MDCT的TDAC功能。在重叠窗口中，这导致Givens旋转，其可以由提升矩阵表示，其中音频信号的时间离散采样值可以首先在成对的基础上相加以构建向量以便顺序地提供与提升矩阵。在通过提升矩阵对向量进行每次乘法之后，执行舍入步骤，使得在输出侧仅将导致整数。通过用整数变换变换窗口整数采样值，可以获得具有整数频谱值的频谱表示。具有逆旋转矩阵和对应的反提升矩阵的逆映射导致精确重建。

10. 发明授权

US06826525B2 Method and device for detecting a transient in a discrete-time audio signal 有权
标题翻译：用于检测离散时间音频信号中的瞬变的方法和装置
公开(公告)号：US06826525B2
公开(公告)日：2004-11-30
申请号：US10183139
申请日：2002-06-25
申请人： Johannes Hilpert , Jürgen Herre , Bernhard Grill , Rainer Buchta , Karlheinz Brandenburg , Heinz Gerhäuser
发明人： Johannes Hilpert , Jürgen Herre , Bernhard Grill , Rainer Buchta , Karlheinz Brandenburg , Heinz Gerhäuser
IPC分类号： G10L1900
CPC分类号： H04B1/665
摘要： A method for detecting a transient in a discrete-time audio signal is performed completely in the time domain and includes the step of segmenting the discrete-time audio signal so as to generate consecutive segments of the same length with unfiltered discrete-time audio signals xs(T−1). The discrete-time audio signal in a current segment is subsequently filtered. Then either the energy of the filtered discrete-time audio signal in the current segment can be compared with the energy of the filtered discrete-time audio signal in a preceding segment or a current relationship between the energy of the filtered discrete-time audio signal in the current segment and the energy of the unfiltered discrete-time audio signal in the current segment can be formed and this current relationship compared with a preceding corresponding relationship. On the basis of the one and/or the other of these comparisons it is detected whether a transient is present in the discrete-time audio signal.
摘要翻译：用于检测离散时间音频信号中的瞬态的方法在时域中完全执行，并且包括分段离散时间音频信号以便生成具有未滤波的离散时间音频信号xs的相同长度的连续片段的步骤（T-1）。随后过滤当前片段中的离散时间音频信号。然后可以将当前段中滤波的离散时间音频信号的能量与前一段中滤波的离散时间音频信号的能量或滤波后的离散时间音频信号的能量之间的当前关系进行比较可以形成当前段的当前段和未过滤离散时间音频信号的能量，并将该当前关系与先前的对应关系进行比较。基于这些比较中的一个和/或另一个，检测离散时间音频信号中是否存在瞬态。

你已经成功收藏专利！

检索式保存成功!

IPRDB

热门服务

关于我们

友情链接

联系方式