公开(公告)号：US20090076801A1

公开(公告)日：2009-03-19

申请号：US12238365

申请日：2008-09-25

申请人： Christian Neubauer ,  Juergen Herre ,  Karlheinz Brandenburg ,  Eric Allamanche

发明人： Christian Neubauer ,  Juergen Herre ,  Karlheinz Brandenburg ,  Eric Allamanche

IPC分类号： G10L19/00

CPC分类号： H04H20/31 ,  G10L19/018 ,  G10L19/02 ,  H04B1/707

摘要： An inventive method for introducing information into a data stream including data about spectral values representing a short-term spectrum of an audio signal first performs a processing of the data stream to obtain the spectral values of the short-term spectrum of the audio signal. Apart from that, the information to be introduced are combined with a spread sequence to obtain a spread information signal, whereupon a spectral representation of the spread information is generated which will then be weighted with an established psychoacoustic maskable noise energy to generate a weighted information signal, wherein the energy of the introduced information is substantially equal to or below the psychoacoustic masking threshold. The weighted information signal and the spectral values of the short-term spectrum of the audio signal will then be summed and afterwards processed again to obtain a processed data stream including both audio information and information to be introduced. By the fact that the information to be introduced are introduced into the data stream without changing to the time domain, the block rastering underlying the short-term spectrum will not be touched, so that introducing a watermark will not lead to tandem encoding effects.

公开(公告)号：US07275036B2

公开(公告)日：2007-09-25

申请号：US10966780

申请日：2004-10-15

申请人： Ralf Geiger ,  Thomas Sporer ,  Karlheinz Brandenburg ,  Juergen Herre ,  Juergen Koller ,  Joachim Deguara

发明人： Ralf Geiger ,  Thomas Sporer ,  Karlheinz Brandenburg ,  Juergen Herre ,  Juergen Koller ,  Joachim Deguara

IPC分类号： G10L19/00

CPC分类号： G10L19/16 ,  G10L19/022 ,  G10L19/20 ,  G10L2019/0005

摘要： A time-discrete audio signal is processed to provide a quantization block with quantized spectral values. Furthermore, an integer spectral representation is generated from the time-discrete audio signal using an integer transform algorithm. The quantization block having been generated using a psychoacoustic model is inversely quantized and rounded to then form a difference between the integer spectral values and the inversely quantized rounded spectral values. The quantization block alone provides a lossy psychoacoustically coded/decoded audio signal after the decoding, whereas the quantization block, together with the combination block, provides a lossless or almost lossless coded and again decoded audio signal in the decoding. By generating the differential signal in the frequency domain, a simpler coder/decoder structure results.

摘要翻译： 处理时间离散音频信号以向量化块提供量化的频谱值。 此外，使用整数变换算法从时间离散音频信号生成整数谱表示。 已经使用心理声学模型产生的量化块被逆量化并舍入，从而形成整数频谱值和逆量化的舍入频谱值之间的差。 量化块单独提供在解码之后的有损心理声学编码/解码音频信号，而量化块与组合块一起在解码中提供无损或几乎无损的编码和再次解码的音频信号。 通过在频域中产生差分信号，可以得到更简单的编码器/解码器结构。

公开(公告)号：US20060147074A1

公开(公告)日：2006-07-06

申请号：US10548667

申请日：2004-03-11

申请人： Berthold Schlenker ,  Thomas Sporer ,  Karlheinz Brandenburg

发明人： Berthold Schlenker ,  Thomas Sporer ,  Karlheinz Brandenburg

IPC分类号： H04R1/02 ,  H04R9/06

CPC分类号： H04R1/025 ,  H04R1/403 ,  H04R2201/021

摘要： A loudspeaker system including a plurality of sonic converters fixed in or to the rear of sound passage openings of a carrier plate is disclosed. The carrier plate is configured in the form of a wall plate for interior spaces or outside facades of buildings and the sonic converters are integrated into the carrier plate. The loudspeaker system or loudspeaker wall plate, respectively, permit the realization of an optimum acoustic irradiation of a space without aesthetically annoying loudspeaker boxes.

公开(公告)号：US07024259B1

公开(公告)日：2006-04-04

申请号：US09889697

申请日：1999-12-15

申请人： Thomas Sporer ,  Roland Bitto ,  Karlheinz Brandenburg

发明人： Thomas Sporer ,  Roland Bitto ,  Karlheinz Brandenburg

IPC分类号： G06F17/00 ,  H04R5/02

CPC分类号： H04S7/30 ,  H04S2420/01

摘要： A system for evaluating the quality of an audio test signal derived from an audio reference signal by coding and decoding, said audio test signal and said audio reference signal each comprising a plurality of channels, comprises a unit for converting the audio reference signal into a first audio reference sum signal at a first reference point and into a second audio reference sum signal at a second reference point and for converting the audio test signal into a first audio test sum signal at the first reference point and into a second audio test sum signal at the second reference point, the audio reference sum signals and the audio test sum signals at the first and second reference points being a superposition of the respective channels, which can be emitted by a plurality of loudspeakers, weighted with a respective transfer function between the respective loudspeaker and the reference point in question, and a unit for evaluating the quality of the audio test sum signals while taking into consideration the audio reference sum signals so as to provide an indication of the quality of the audio test signal. The system according to the present invention permits real rooms and an arbitrary number of channels of the audio test signal to be taken into account so as to execute a listening-adapted evaluation of the quality of a specific coding/decoding method.

摘要翻译： 一种用于评估通过编码和解码从音频参考信号导出的音频测试信号的质量的系统，所述音频测试信号和每个包括多个通道的所述音频参考信号包括用于将音频参考信号转换成第一 在第一参考点处的音频参考和信号和在第二参考点处的第二音频参考和信号，并且用于将音频测试信号转换成第一参考点处的第一音频测试和信号并转换成第二音频测试和信号 第二参考点，第一参考点和第二参考点处的音频参考和信号和音频测试和信号是可由多个扬声器发射的相应通道的叠加，其由相应的传递函数加权 扬声器和参考点，以及用于评估音频测试和信号质量的单元 以考虑音频参考和信号，以便提供音频测试信号的质量的指示。 根据本发明的系统允许考虑音频测试信号的实际房间和任意数量的信道，以便执行对特定编码/解码方法的质量的倾听适应评估。

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。

公开(公告)号：US06185539B2

公开(公告)日：2001-02-06

申请号：US09077395

申请日：1998-05-26

申请人： Oliver Kunz ,  Martin Dietz ,  Rainer Buchta ,  Jürgen Zeller ,  Karlheinz Brandenburg ,  Martin Sieler ,  Heinz Gerhäuser

发明人： Oliver Kunz ,  Martin Dietz ,  Rainer Buchta ,  Jürgen Zeller ,  Karlheinz Brandenburg ,  Martin Sieler ,  Heinz Gerhäuser

IPC分类号： H04B166

CPC分类号： H04B1/665

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。

公开(公告)号：USRE36714E

公开(公告)日：2000-05-23

申请号：US622313

申请日：1994-11-10

申请人： Karlheinz Brandenburg ,  James David Johnston

发明人： Karlheinz Brandenburg ,  James David Johnston

IPC分类号： G10L7/04

摘要： A method is disclosed for determining estimates of the perceived noise masking level of audio signals as a function of frequency. By developing a randomness metric related to the euclidian distance between (i) actual frequency components amplitude and phase for each block of sampled values of the signal and (ii) predicted values for these components based on values in prior blocks, it is possible to form a tonality index which provides more detailed information useful in forming the noise masking function. Application of these techniques is illustrated in a coding and decoding context for audio recording or transmission. The noise spectrum is shaped based on a noise threshold and a tonality measure for each critical frequency-band (bark).

摘要翻译： 公开了一种用于确定作为频率的函数的音频信号的感知噪声屏蔽水平的估计的方法。 通过开发与（i）信号的采样值的每个块的实际频率分量振幅和相位之间的欧几里德距离相关的随机度量，以及（ii）基于先前块中的值的这些分量的预测值，可以形成 一种音调指数，其提供了用于形成噪声屏蔽功能的更详细的信息。 这些技术的应用在用于音频记录或传输的编码和解码环境中被说明。 基于每个关键频带（树皮）的噪声阈值和音调测量，噪声谱是成形的。

公开(公告)号：US08117027B2

公开(公告)日：2012-02-14

申请号：US12238365

申请日：2008-09-25

申请人： Christian Neubauer ,  Juergen Herre ,  Karlheinz Brandenburg ,  Eric Allamanche

发明人： Christian Neubauer ,  Juergen Herre ,  Karlheinz Brandenburg ,  Eric Allamanche

IPC分类号： G10L19/02 ,  H04B1/66 ,  H04B1/69

CPC分类号： H04H20/31 ,  G10L19/018 ,  G10L19/02 ,  H04B1/707

摘要： Techniques for introducing information into a data stream first obtains the spectral values of the short-term spectrum of the audio signal. Separately, information to be introduced are combined with a spread sequence obtaining a spread information signal, whereupon a spectral representation of the spread information is generated, then weighted with an established psychoacoustic maskable noise energy to generate a weighted information signal, wherein energy of the introduced information is substantially equal to or below the psychoacoustic masking threshold. The weighted information signal and the spectral values of the short-term spectrum of the audio signal are then summed and afterwards processed again to obtain a processed data stream including audio information and information to be introduced. Because the information to be introduced are introduced without changing to the time domain, the block rastering underlying the short-term spectrum are not touched, thus introducing a watermark will not lead to tandem encoding effects.

摘要翻译： 将信息引入数据流的技术首先获得音频信号的短期频谱的频谱值。 单独地，要引入的信息与获得扩展信息信号的扩展序列组合，从而生成扩展信息的频谱表示，然后用已建立的心理声学可屏蔽噪声能量进行加权，以产生加权信息信号，其中引入的能量 信息基本上等于或低于心理声学屏蔽阈值。 然后，将加权信息信号和音频信号的短期频谱的频谱值相加，然后再次进行处理，以获得包括音频信息和要引入的信息的处理数据流。 由于在不改变时域的情况下引入要引入的信息，因此不会触及短期频谱下的块划像，因此引入水印不会导致串联编码效果。

公开(公告)号：US07454327B1

公开(公告)日：2008-11-18

申请号：US10089950

申请日：2000-10-05

申请人： Christian Neubauer ,  Juergen Herre ,  Karlheinz Brandenburg ,  Eric Allamanche

发明人： Christian Neubauer ,  Juergen Herre ,  Karlheinz Brandenburg ,  Eric Allamanche

IPC分类号： G10L19/02 ,  H04B1/69

CPC分类号： H04H20/31 ,  G10L19/018 ,  G10L19/02 ,  H04B1/707

摘要： An inventive method for introducing information into a data stream including data about spectral values representing a short-term spectrum of an audio signal first performs a processing of the data stream to obtain the spectral values of the short-term spectrum of the audio signal. Apart from that, the information to be introduced are combined with a spread sequence to obtain a spread information signal, whereupon a spectral representation of the spread information is generated which will then be weighted with an established psychoacoustic maskable noise energy to generate a weighted information signal, wherein the energy of the introduced information is substantially equal to or below the psychoacoustic masking threshold. The weighted information signal and the spectral values of the short-term spectrum of the audio signal will then be summed and afterwards processed again to obtain a processed data stream including both audio information and information to be introduced. By the fact that the information to be introduced are introduced into the data stream without changing to the time domain, the block rastering underlying the short-term spectrum will not be touched, so that introducing a watermark will not lead to tandem encoding effects.

摘要翻译： 将信息引入包括表示音频信号的短期频谱的频谱值的数据的数据流的创新方法首先执行数据流的处理以获得音频信号的短期频谱的频谱值。 除此之外，将要引入的信息与扩展序列组合以获得扩展信息信号，从而生成扩展信息的频谱表示，然后将其利用已建立的心理声学可屏蔽噪声能量进行加权，以产生加权信息信号 ，其中所引入的信息的能量基本上等于或低于心理声学掩蔽阈值。 然后将加权信息信号和音频信号的短期频谱的频谱值相加，然后再次处理，以获得包括音频信息和要引入的信息的处理数据流。 由于将要引入的信息被引入到数据流而不改变到时域的事实，所以短期频谱下面的块划像不会被触及，所以引入水印不会导致串联编码效应。

公开(公告)号：US20080206731A1

公开(公告)日：2008-08-28

申请号：US11995563

申请日：2006-09-06

申请人： Fanny Bastianova-Klett ,  Karlheinz Brandenburg

发明人： Fanny Bastianova-Klett ,  Karlheinz Brandenburg

IPC分类号： G09B7/00

CPC分类号： G09B7/00

摘要： An apparatus for compiling a test comprises a database having a plurality of test tasks stored therein, each test task being associated with a task type, means for selecting test tasks from the database to obtain a multitude of selected test tasks, and means for outputting the selected test tasks of the test to a user. The means for selecting test tasks comprises means for selecting, for a task type, at least one test task from the database and for taking the selected test task over to the multitude of selected test tasks if a test task for the task type is available in the database, and an exception-handling logic configured to search the database, for a task type for which no test task is available in the database, for a replacement test task according to a given replacement rule and take same over to the multitude of selected tasks.

摘要翻译： 用于编译测试的装置包括具有存储在其中的多个测试任务的数据库，每个测试任务与任务类型相关联，用于从数据库中选择测试任务以获得大量所选择的测试任务的装置，以及用于输出 所选测试任务的测试给用户。 用于选择测试任务的手段包括用于为任务类型选择来自数据库的至少一个测试任务并且用于将所选择的测试任务转移到多个所选择的测试任务的装置，如果任务类型的测试任务可用于 该数据库和一个异常处理逻辑被配置为根据给定的替换规则为替换测试任务针对数据库中没有测试任务可用的任务类型搜索数据库，并将其与多个所选择的 任务。