公开(公告)号：US09397771B2

公开(公告)日：2016-07-19

申请号：US13333461

申请日：2011-12-21

申请人： Peter Jax ,  Johann-Markus Batke ,  Johannes Boehm ,  Sven Kordon

发明人： Peter Jax ,  Johann-Markus Batke ,  Johannes Boehm ,  Sven Kordon

IPC分类号： H04R5/00 ,  H04H20/89 ,  G10L19/008

CPC分类号： H04H20/89 ,  G10L19/008

摘要： Representations of spatial audio scenes using higher-order Ambisonics HOA technology typically require a large number of coefficients per time instant. This data rate is too high for most practical applications that require real-time transmission of audio signals. According to the invention, the compression is carried out in spatial domain instead of HOA domain. The (N+1)2 input HOA coefficients are transformed into (N+1)2 equivalent signals in spatial domain, and the resulting (N+1)2 time-domain signals are input to a bank of parallel perceptual codecs. At decoder side, the individual spatial-domain signals are decoded, and the spatial-domain coefficients are transformed back into HOA domain in order to recover the original HOA representation.

摘要翻译： 使用高阶Ambisonics HOA技术的空间音频场景的表示通常每个时刻需要大量的系数。 对于需要实时传输音频信号的大多数实际应用，此数据速率太高。 根据本发明，压缩在空间域中而不是HOA域进行。 （N + 1）2个输入HOA系数在空间域中变换为（N + 1）2个等效信号，并将得到的（N + 1）2个时域信号输入到一组并行感知编解码器。 在解码器侧，对各个空间域信号进行解码，并将空间域系数变换回到HOA域，以恢复原始的HOA表示。

公开(公告)号：US09338574B2

公开(公告)日：2016-05-10

申请号：US14130074

申请日：2012-06-15

申请人： Peter Jax ,  Johann-Markus Batke

发明人： Peter Jax ,  Johann-Markus Batke

IPC分类号： H04S5/00 ,  G10L21/00 ,  H04S3/00

CPC分类号： H04S5/00 ,  G10L21/00 ,  H04R2205/024 ,  H04S3/002 ,  H04S2400/11 ,  H04S2420/11

摘要： Higher-order Ambisonics HOA is a representation of spatial sound fields that facilitates capturing, manipulating, recording, transmission and playback of complex audio scenes with superior spatial resolution, both in 2D and 3D. The sound field is approximated at and around a reference point in space by a Fourier-Bessel series. The invention uses space warping for modifying the spatial content and/or the reproduction of sound-field information that has been captured or produced as a higher-order Ambisonics representation. Different warping characteristics are feasible for 2D and 3D sound fields. The warping is performed in space domain without performing scene analysis or decomposition. Input HOA coefficients with a given order are decoded to the weights or input signals of regularly positioned (virtual) loudspeakers.

摘要翻译： 高阶环境HOA是空间声场的表示，有助于在2D和3D中捕获，操纵，记录，传输和播放具有优异空间分辨率的复杂音频场景。 声场通过傅里叶Bessel系列在空间中的参考点附近和周围近似。 本发明使用空间扭曲来修改作为高阶Ambisonics表示被捕获或产生的声场信息的空间内容和/或再现。 不同的翘曲特性对于2D和3D声场是可行的。 翘曲在空间域执行，而不进行场景分析或分解。 具有给定次序的输入HOA系数被解码为规则定位（虚拟）扬声器的权重或输入信号。

公开(公告)号：US08612216B2

公开(公告)日：2013-12-17

申请号：US12223362

申请日：2006-01-31

申请人： Martin Gartner ,  Bernd Geiser ,  Peter Jax ,  Stefan Schandl ,  Herve Taddei ,  Peter Vary

发明人： Martin Gartner ,  Bernd Geiser ,  Peter Jax ,  Stefan Schandl ,  Herve Taddei ,  Peter Vary

IPC分类号： G10L19/08 ,  G10L19/12

CPC分类号： G10L19/0208 ,  G10L21/038

摘要： To form an audio signal, frequency components of the audio signal which are allotted to a first subband are formed by means of a subband decoder using supplied fundamental period values which respectively indicate a fundamental period for the audio signal. Frequency components of the audio signal which are allotted to a second subband are formed by exciting an audio synthesis filter using an excitation signal which is specific to the second subband. To produce this excitation signal, an excitation signal generator derives a fundamental period parameter from the fundamental period values. The fundamental period parameter is used by the excitation signal generator to form pulses with a pulse shape which is dependent on the fundamental period parameter at an interval of time which is determined by the fundamental period parameter and to mix them with a noise signal.

摘要翻译： 为了形成音频信号，分配给第一子带的音频信号的频率分量通过使用提供的基本周期值的子带解码器形成，该基带周期值分别表示音频信号的基本周期。 通过使用特定于第二子带的激励信号激励音频合成滤波器来形成分配给第二子带的音频信号的频率分量。 为了产生该激励信号，激励信号发生器从基本周期值导出基本周期参数。 基频周期参数由激励信号发生器使用，以形成脉冲形状的脉冲，脉冲形状取决于基本周期参数，该时间间隔由基本周期参数决定，并将其与噪声信号混合。

公开(公告)号：US08428942B2

公开(公告)日：2013-04-23

申请号：US12227189

申请日：2007-05-12

申请人： Peter Jax ,  Oliver Wuebbolt ,  Johannes Boehm

发明人： Peter Jax ,  Oliver Wuebbolt ,  Johannes Boehm

IPC分类号： G10L19/00 ,  G10L19/02

CPC分类号： G10L19/24 ,  G10L19/0204 ,  G10L19/173

摘要： At the time of encoding audio content, the finally required data rate for delivery to the customer may be unknown. A data format is disclosed that is optimized for serving as Intermediate Format for efficient and fast recoding, to obtain one or more standard complying lossy encoded data streams with flexible data rates. Encoding can be performed in two steps that are inter-coordinated for cooperating, but may be locally and/or temporally separate. Between the partial encoders encoding parameters and/or auxiliary data are transmitted in a separate parameter enhancement layer, which complements a lossy data stream and can be used by the second encoder or transcoder for fast and computationally efficient implementation of the second encoding step. An additional lossless enhancement layer allows lossless reconstruction.

摘要翻译： 在对音频内容进行编码时，传送给客户的最终所需的数据速率可能是未知的。 公开了针对用于高效和快速重新编码的中间格式进行优化的数据格式，以获得具有灵活数据速率的一个或多个标准符合有损编码数据流。 编码可以在两个步骤中执行，这两个步骤是协调的，用于协作，但可以在本地和/或时间上分离。 编码参数和/或辅助数据的部分编码器之间的传输是在单独的参数增强层中传输的，其补充有损数据流，并可被第二编码器或代码转换器用于快速和计算上有效地实现第二编码步骤。 额外的无损增强层允许无损重建。

公开(公告)号：US08135584B2

公开(公告)日：2012-03-13

申请号：US12223359

申请日：2006-01-31

申请人： Bernd Geiser ,  Peter Jax ,  Stefan Schandl ,  Herve Taddei

发明人： Bernd Geiser ,  Peter Jax ,  Stefan Schandl ,  Herve Taddei

IPC分类号： G10L19/00

CPC分类号： G10L21/038 ,  G10L19/12 ,  G10L2019/0001

摘要： According to the invention, an excitation signal is generated as a result of sampled excitation values in order to excite an audio synthesis filter, the generated sampled excitation values being continuously stored in an adaptive codebook. A noise generator is provided which continuously generates random sampled values. A sequence of the stored sampled excitation values is selected from the adaptive codebook based on a fed audio fundamental frequency parameter by means of which a time gap between the sequence that is to be selected and the actual time reference is predefined. The excitation signal is generated by mixing the selected sequence with a random sequence encompassing actual random sampled valued of the noise generator.

摘要翻译： 根据本发明，作为采样激励值的结果产生激励信号，以激发音频合成滤波器，所生成的采样激励值被连续存储在自适应码本中。 提供连续产生随机采样值的噪声发生器。 基于馈送的音频基频参数，从自适应码本中选择存储的采样激励值的序列，通过该序列，预定要选择的序列与实际时间基准之间的时间间隔。 通过将所选择的序列与包含噪声发生器的实际随机采样值的随机序列混合来产生激励信号。

公开(公告)号：US20100126250A1

公开(公告)日：2010-05-27

申请号：US12696450

申请日：2010-01-29

申请人： Peter Jax

发明人： Peter Jax

IPC分类号： G01M3/28

CPC分类号： G01M3/222

摘要： In a method for detecting a leak in a double pipe, a medium located in an intermediate chamber between the interior and exterior pipe is moved toward a first end of the pipe, whereupon the medium flows in from the second end. On the first end the medium is examined for any leakage of a characteristic material, upon the detection of which a leakage signal is generated and a location of the leakage is calculated based on the transport time of the characteristic material from the leakage to the first end and on the mass flow of the medium. A corresponding device contains a conveyor unit for moving the medium through the intermediate chamber. A material sensor is disposed at the first end for examining the medium for the characteristic material. A control and analysis unit is provided for generating a leakage signal and calculating the location of the leakage.

摘要翻译： 在用于检测双管中的泄漏的方法中，位于内管和外管之间的中间室中的介质朝向管的第一端移动，于是介质从第二端流入。 在第一端，检查介质的特征材料的任何泄漏，在检测到产生泄漏信号并基于特征材料从泄漏到第一端的运输时间计算泄漏的位置时 以及介质的质量流量。 相应的装置包含用于通过中间室移动介质的输送单元。 材料传感器设置在第一端，用于检查特征材料的介质。 提供控制和分析单元，用于产生泄漏信号并计算泄漏的位置。

公开(公告)号：US20090164226A1

公开(公告)日：2009-06-25

申请号：US12226992

申请日：2007-04-18

申请人： Johannes Boehm ,  Peter Jax ,  Florian Keiler ,  Oliver Wuebbolt ,  Sven Kordon

发明人： Johannes Boehm ,  Peter Jax ,  Florian Keiler ,  Oliver Wuebbolt ,  Sven Kordon

IPC分类号： G10L19/04 ,  G10L19/00

CPC分类号： G10L19/24 ,  G10L19/0017

摘要： In lossy based lossless coding a PCM audio signal passes through a lossy encoder to a lossy decoder. The lossy encoder provides a lossy bit stream. The difference signal between the PCM signal and the lossy decoder output is lossless encoded, providing an extension bit stream. The invention facilitates enhancing a lossy perceptual audio encoding/decoding by an extension that enables mathematically exact reproduction of the original waveform using enhanced de-correlation, and provides additional data for reconstructing at decoder site an intermediate-quality audio signal. The lossless extension can be used to extend the widely used mp3 encoding/decoding to lossless encoding/decoding and superior quality mp3 encoding/de-coding.

摘要翻译： 在基于有损耗的无损编码中，PCM音频信号通过有损编码器到有损解码器。 有损编码器提供有损比特流。 PCM信号和有损解码器输出之间的差分信号是无损编码的，提供扩展位流。 本发明有助于通过扩展来增强有损感知音频编码/解码，其使得能够使用增强的去相关来数学地精确地再现原始波形，并且提供用于在解码器位置重建中等质量音频信号的附加数据。 无损扩展可用于将广泛使用的mp3编码/解码扩展到无损编码/解码以及优质的mp3编码/解码。

公开(公告)号：US5215409A

公开(公告)日：1993-06-01

申请号：US730619

申请日：1991-07-16

申请人： Peter Jax ,  Klaus Ruthrof ,  Karl-Heinz Bode ,  Klaus Gunther

发明人： Peter Jax ,  Klaus Ruthrof ,  Karl-Heinz Bode ,  Klaus Gunther

IPC分类号： B09B1/00 ,  E02D31/00 ,  G01M3/02 ,  G01M3/20

CPC分类号： B09B1/00 ,  E02D31/00 ,  G01M3/02 ,  G01M3/20

摘要： A device for sealing off and monitoring a volume includes two seals. Support elements space the seals apart from one another. At least one conduit is disposed between the seals and the support elements. The at least one conduit has an inlet opening and an outlet opening for a medium. At least one sensor is associated with the outlet line. The two seals have edges being tightly joined to one another leaving the inlet and outlet openings of the at least one conduit free. The support elements are formed of a flow-hindering material. At least two devices may be tightly joined together to form a field, with the outlet opening of the at least one conduit of one of the at least two devices being connected to the inlet opening of the at least one conduit of another of the at least two devices.

摘要翻译： 用于密封和监测体积的装置包括两个密封件。 支撑元件将密封件彼此分开。 至少一个导管设置在密封件和支撑元件之间。 至少一个导管具有用于介质的入口和出口。 至少一个传感器与出口管线相关联。 两个密封件具有彼此紧密接合的边缘，留下至少一个导管的入口和出口开口自由。 支撑元件由阻流材料形成。 至少两个装置可以紧密连接在一起以形成一个场，其中至少两个装置中的一个的至少一个管道的出口开口连接到至少另一个装置的至少一个导管的入口 两个设备。

公开(公告)号：US09093065B2

公开(公告)日：2015-07-28

申请号：US12311129

申请日：2007-09-06

申请人： Peter Jax ,  Sven Kordon

发明人： Peter Jax ,  Sven Kordon

IPC分类号： G10L19/00 ,  G10L25/00 ,  H04B1/66 ,  G10L19/022 ,  G10L19/16 ,  G06F3/16 ,  G10L19/008 ,  G10L19/005

CPC分类号： G10L19/022 ,  G06F3/16 ,  G10L19/005 ,  G10L19/008 ,  G10L19/173 ,  H04B1/66

摘要： The present invention provides method and device for transcoding between audio coding formats with different time-frequency analysis domains, as used for example by MP EG-AAC and mp3, particularly for facilitated and faster transcoding between such audio signals. A method for transcoding a framed audio signal from a first parameter domain into a second parameter domain comprises linearly transforming two or more parameters of the first parameter domain to at least one parameter of the second parameter domain, wherein the two or more parameters of the first parameter domain come from different frames of the audio signal in the first parameter domain. The linear transformation can be described as a matrix and implemented as a look-up table, wherein the matrix transformation coefficients below -60 Decibels are neglected.

摘要翻译： 本发明提供了用于例如通过MP EG-AAC和mp3所使用的不同时间 - 频率分析域的音频编码格式之间进行代码转换的方法和装置，特别是用于在这种音频信号之间促进和更快的代码转换。 将成帧音频信号从第一参数域转码成第二参数域的方法包括将第一参数域的两个或多个参数线性变换成第二参数域的至少一个参数，其中第一参数域的两个或多个参数 参数域来自第一参数域中音频信号的不同帧。 线性变换可以被描述为矩阵并被实现为查找表，其中低于-60分贝的矩阵变换系数被忽略。

公开(公告)号：US08503535B2

公开(公告)日：2013-08-06

申请号：US12734625

申请日：2008-11-10

申请人： Peter Jax

发明人： Peter Jax

IPC分类号： H04N11/02 ,  H04N11/04 ,  H04N7/12 ,  H04N7/32 ,  G10L19/00 ,  G10L19/02

CPC分类号： G10L19/0212 ,  G10L19/0017 ,  G10L19/24 ,  G10L19/26 ,  H04N19/117 ,  H04N19/136 ,  H04N19/46 ,  H04N19/60 ,  H04N19/80 ,  H04N19/85

摘要： An integer-reversible MDCT transformation is split into consecutive lifting steps, each introducing considerable rounding errors to the signal. Without noise shaping the rounding error noise will impact all frequency bins of the transformed signal equally. This is a particular problem for low signal level frequency bins. The invention limits the impact of rounding error noise coming with each lifting step in the integer-reversible transformation on the data rate of a lossless audio codec. The filter coefficients of an adaptive noise shaping filter for transform coefficients are adapted in individual lifting steps according to the current time domain signal characteristics. As an alternative, an auto-regressive pre-filter can be added in front of the lossless transformation, for raising the level of frequency regions with low power to decrease the dominance of rounding errors in these areas. Both processes can be combined to further improve lossless codec compression ratio.

摘要翻译： 整数可逆MDCT变换被分为连续的提升步骤，每个引入相当大的舍入误差给信号。 无噪声整形时，舍入误差将对变换后的信号的所有频率段进行平均的影响。 这对于低信号电平频率箱来说是一个特殊的问题。 本发明限制了对于无损音频编解码器的数据速率的整数可逆变换中每个提升步长的舍入误差噪声的影响。 用于变换系数的自适应噪声整形滤波器的滤波器系数根据当前时域信号特性在各个提升步骤中进行调整。 作为替代，可以在无损变换之前添加自回归预滤波器，以提高具有低功率的频率区域的水平，以减少这些区域中舍入误差的优势。 这两个过程可以组合起来，进一步提高无损编解码压缩率。