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    • 31. 发明授权
    • System and method for evaluating the quality of multi-channel audio signals
    • 用于评估多声道音频信号质量的系统和方法
    • US07024259B1
    • 2006-04-04
    • US09889697
    • 1999-12-15
    • Thomas SporerRoland BittoKarlheinz Brandenburg
    • Thomas SporerRoland BittoKarlheinz Brandenburg
    • G06F17/00H04R5/02
    • H04S7/30H04S2420/01
    • 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.
    • 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。
    • 32. 再颁专利
    • Perceptual coding of audio signals
    • 音频信号的感知编码
    • USRE36714E
    • 2000-05-23
    • US622313
    • 1994-11-10
    • Karlheinz BrandenburgJames David Johnston
    • Karlheinz BrandenburgJames David Johnston
    • G10L7/04
    • 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).
    • 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。
    • 33. 发明授权
    • Method and apparatus for introducing information into a data stream and method and apparatus for encoding an audio signal
    • 用于将信息引入数据流的方法和装置以及用于对音频信号进行编码的方法和装置
    • US08117027B2
    • 2012-02-14
    • US12238365
    • 2008-09-25
    • Christian NeubauerJuergen HerreKarlheinz BrandenburgEric Allamanche
    • Christian NeubauerJuergen HerreKarlheinz BrandenburgEric Allamanche
    • G10L19/02H04B1/66H04B1/69
    • H04H20/31G10L19/018G10L19/02H04B1/707
    • 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.
    • 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。
    • 34. 发明授权
    • Method and apparatus for introducing information into a data stream and method and apparatus for encoding an audio signal
    • 用于将信息引入数据流的方法和装置以及用于对音频信号进行编码的方法和装置
    • US07454327B1
    • 2008-11-18
    • US10089950
    • 2000-10-05
    • Christian NeubauerJuergen HerreKarlheinz BrandenburgEric Allamanche
    • Christian NeubauerJuergen HerreKarlheinz BrandenburgEric Allamanche
    • G10L19/02H04B1/69
    • H04H20/31G10L19/018G10L19/02H04B1/707
    • 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.
    • 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。
    • 35. 发明申请
    • Apparatus, Method and Computer Program for Compiling a Test as Well as Apparatus, Method and Computer Program for Testing an Examinee
    • 用于测试的装置,方法和计算机程序以及用于测试检查者的装置,方法和计算机程序
    • US20080206731A1
    • 2008-08-28
    • US11995563
    • 2006-09-06
    • Fanny Bastianova-KlettKarlheinz Brandenburg
    • Fanny Bastianova-KlettKarlheinz Brandenburg
    • G09B7/00
    • G09B7/00
    • 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.
    • 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。
    • 37. 发明授权
    • Method for converting a music signal into a note-based description and for referencing a music signal in a data bank
    • 用于将音乐信号转换为基于音符的描述并用于参考数据库中的音乐信号的方法
    • US07064262B2
    • 2006-06-20
    • US10473462
    • 2002-04-04
    • Frank KlefenzKarlheinz BrandenburgMatthias Kaufmann
    • Frank KlefenzKarlheinz BrandenburgMatthias Kaufmann
    • G10H7/00
    • G10H1/0041
    • In a method for transferring a music signal into a note-based description, a frequency-time representation of the music signal is first generated, the frequency-time representation comprising coordinate tuples, a coordinate tuple including a frequency value and a time value, the time value indicating the time of occurrence of the assigned frequency in the music signal. Thereupon, a fit function will be calculated as a function of the time, the course of which is determined by the coordinate tuples of the frequency-time representation. For time-segmenting the frequency-time representation, at least two adjacent extreme values of the fit function will be determined. On the basis of the determined extreme values, a segmenting will be carried out, a segment being limited by two adjacent extreme values of the fit function, the time length of the segments indicating a time length of a note for the segment. For pitch determination, a pitch for the segment using coordinate tuples in the segment will be determined. For calculating the fit function and determining extreme values of the fit function for segmenting, no requirements are made to the music signal which is to be transferred into a note-based representation. The method is thus also suitable for continuous music signals.
    • 在将音乐信号转换为基于音符的描述的方法中,首先生成音乐信号的频率 - 时间表示,包括坐标元组的频率 - 时间表示,包括频率值和时间值的坐标元组, 指示音乐信号中分配频率发生的时间的时间值。 因此,拟合函数将被计算为时间的函数,其时间由频率 - 时间表示的坐标元组决定。 为了对频率 - 时间表示进行时间分割,将确定拟合函数的至少两个相邻极值。 基于所确定的极值,将执行分段,段被拟合函数的两个相邻极值限制,段的时间长度指示段的音符的时间长度。 对于音调确定,将确定段中使用坐标元组的段的音高。 为了计算拟合函数并确定用于分割的拟合函数的极值,对要转移到基于笔记的表示中的音乐信号没有要求。 因此,该方法也适用于连续的音乐信号。
    • 40. 发明授权
    • Method of coding a plurality of audio signals
    • 编码多个音频信号的方法
    • US5701346A
    • 1997-12-23
    • US704730
    • 1996-09-12
    • Jurgen HerreBerhard GrillErnst EberleinKarlheinz BrandenburgDieter Seitzer
    • Jurgen HerreBerhard GrillErnst EberleinKarlheinz BrandenburgDieter Seitzer
    • H04S5/02H03M7/30H04B14/04H04H20/88H04S3/00H04S3/02
    • H04H20/88H04S3/02
    • In a method of coding a plurality of audio signals, the left and the right basic channel as well as the central channel are combined by joint stereo coding so as to obtain a jointly coded signal, which is decoded so as to provide simulated decoded signals. The simulated decoded signals and two surround channels are combined by matricization by means of a compatibility matrix so as to form compatible signals which are suitable for decoding by existing decoders. In order to avoid audible disturbances caused by excessive energy contents of the compatible signals, which would occur if joint stereo coding and decoding were carried out prior to carrying out the matricization, the compatible signals or the simulated decoded signals are dynamically weighted by means of a dynamic correction factor in such a way that the compatible signals are approximated with regard to their energy to the energy of signals which would be obtained if the two basic channels and the central channel as well as the surround channels were matricized directly.
    • PCT No.PCT / EP95 / 00378 371日期:1996年9月12日 102(e)日期1996年9月12日PCT提交1995年2月2日PCT公布。 公开号WO95 / 26083 日期1995年9月28日在一种对多个音频信号进行编码的方法中,通过联合立体声编码来组合左右基本频道以及中央频道,以获得联合编码的信号,其被解码为 提供模拟解码信号。 模拟的解码信号和两个环绕声道通过借助于兼容性矩阵的校准来组合,以形成适合于现有解码器解码的兼容信号。 为了避免由于兼容信号的能量含量过多引起的可听到的干扰,如果在执行基准化之前进行了联合的立体声编码和解码,则会发生兼容信号或模拟解码信号通过 动态校正因子,使得将兼容信号相对于其能量相对于如果两个基本信道和中央信道以及环绕信道被直接校准而获得的信号的能量近似即可。