Audio reproduction system and method for reproducing an audio signal
    32.
    发明授权
    Audio reproduction system and method for reproducing an audio signal 有权
    用于再现音频信号的音频再现系统和方法

    公开(公告)号:US07706544B2

    公开(公告)日:2010-04-27

    申请号:US11099156

    申请日:2005-04-05

    IPC分类号: H04R5/00 H04R5/02

    摘要: An audio reproduction system is divided into a central wave-field synthesis module and a plurality of loudspeaker modules disposed in a distributed way, wherein synthesis signals for the individual loudspeakers as well as corresponding channel information associated to the synthesis signals are calculated in the central wave-field synthesis module. The synthesis signals for a loudspeaker as well as associated channel information will then be transmitted to respective loudspeaker modules via a transmission path, wherein every loudspeaker module obtains the synthesis signals and associated channel information intended for the loudspeaker associated to the loudspeaker module. A distributed audio rendering and digital/analog converting takes place in the loudspeaker module to generate the actually analog loudspeaker signals in a distributed way in spatial proximity to every loudspeaker. The division into a central wave-field synthesis module and the plurality of distributed loudspeaker modules allows that audio reproduction systems that are scalable with regard to the price can be generated in order to offer systems of different size scalable in price particularly for cinema reproduction rooms varying strongly in size.

    摘要翻译: 音频再现系统被分为中心波场合成模块和以分布式方式布置的多个扬声器模块,其中针对各个扬声器的合成信号以及与合成信号相关联的对应信道信息在中央波中计算 场合成模块。 然后,用于扬声器的合成信号以及相关联的信道信息将经由传输路径发送到相应的扬声器模块,其中每个扬声器模块获得旨在用于与扬声器模块相关联的扬声器的合成信号和相关信道信息。 分布式音频渲染和数字/模拟转换发生在扬声器模块中,以分布式的方式在每个扬声器的空间附近产生实际的模拟扬声器信号。 分为中央波场合成模块和多个分布式扬声器模块允许可以生成关于价格可扩展的音频再现系统,以便提供价格可变的不同尺寸的系统,特别是对于不同的电影再现室 强大的。

    Wave field synthesis apparatus and method of driving an array of loudspeakers
    33.
    发明授权
    Wave field synthesis apparatus and method of driving an array of loudspeakers 有权
    用于驱动扬声器阵列的波场合成装置和方法

    公开(公告)号:US07684578B2

    公开(公告)日:2010-03-23

    申请号:US11305546

    申请日:2005-12-16

    摘要: In a wave field synthesis apparatus for driving an array of loudspeakers with drive signals, the loudspeakers being arranged at different defined positions, a drive signal for a loudspeaker being based on an audio signal associated with a virtual source having a virtual position with reference to the loudspeaker array and on the defined position of the loudspeaker, at first relevant loudspeakers of the loudspeaker array are determined on the basis of the position of the virtual source, a predefined listener position, and the defined positions of the loudspeakers, so that artifacts due to loudspeaker signals moving opposite to a direction from the virtual source to the predefined listener position are reduced. Downstream to means for calculating the drive signal components for the relevant loudspeakers and for a virtual source, there is means for providing the drive signal components for the relevant loudspeakers for the virtual source to the relevant loudspeakers, wherein no drive signals for the virtual source are provided to loudspeakers of the loudspeaker array not belonging to the relevant loudspeakers. With this, artifacts in an area of the audience room due to a generation wave field are suppressed, so that in this area only the useful wave field is heard in artifact-free manner.

    摘要翻译: 在用于驱动具有驱动信号的扬声器阵列的波场合成装置中,扬声器布置在不同的限定位置,扬声器的驱动信号基于与具有虚拟位置的虚拟源相关联的音频信号,该虚拟源参考 扬声器阵列并且在扬声器的限定位置上,在扬声器阵列的第一相关扬声器的基础上,基于虚拟源的位置,预定义的收听者位置以及所定义的扬声器的位置来确定,从而由于 与从虚拟源到预定义的收听者位置的方向相反的扬声器信号被减少。 下游是用于计算相关扬声器和虚拟源的驱动信号分量的装置,存在用于向相关扬声器提供用于虚拟源的相关扬声器的驱动信号分量的装置,其中虚拟源的驱动信号不是 提供给不属于相关扬声器的扬声器阵列的扬声器。 由此,抑制了由于发电波场而在观众室的区域中的伪影,所以在该区域中,仅以无伪影的方式听到有用的波场。

    Loudspeaker
    34.
    发明授权
    Loudspeaker 有权
    喇叭

    公开(公告)号:US07391879B2

    公开(公告)日:2008-06-24

    申请号:US11046123

    申请日:2005-01-28

    IPC分类号: H04R1/00

    摘要: An inventive loudspeaker includes a diaphragm, a first excitation means for generating structure-borne sound in the diaphragm, and a second excitation means, different from the first one, for setting the diaphragm into a longitudinal vibrational motion in a direction perpendicular to the extension of the diaphragm. In accordance with the invention, the problem of insufficient bass reproduction and/or of the magnitude conflicting with invisible integration or installation is solved in that a second exciter system is introduced, which uniformly moves the diaphragm, or the plate serving as the diaphragm, forward and backward in addition to the bending waves of the structure-borne sound. The sound reproduction therefore is possible across the entire audio-frequency range without impeding the goal of invisible integration or installation.

    摘要翻译: 本发明的扬声器包括隔膜,用于在隔膜中产生结构声音的第一激励装置和与第一激励装置不同于第一激励装置的第二激励装置,用于将振膜设置成垂直于垂直于 隔膜。 根据本发明,解决了将不正确的低音再现和/或与不可见的整合或安装相冲突的大小的问题引入到第二激励器系统中,其将隔膜或用作隔膜的板均匀地移动 并且除了结构声音的弯曲波之外,还有向后的方向。 因此,声音再现在整个音频范围内是可能的,而不会妨碍不可见的整合或安装的目的。

    Apparatus and method for coding a time-discrete audio signal to obtain coded audio data and for decoding coded audio data
    35.
    发明授权
    Apparatus and method for coding a time-discrete audio signal to obtain coded audio data and for decoding coded audio data 有权
    用于编码时分离音频信号以获得编码音频数据和解码编码音频数据的装置和方法

    公开(公告)号:US07275036B2

    公开(公告)日:2007-09-25

    申请号:US10966780

    申请日:2004-10-15

    IPC分类号: G10L19/00

    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。

    System and method for evaluating the quality of multi-channel audio signals
    37.
    发明授权
    System and method for evaluating the quality of multi-channel audio signals 有权
    用于评估多声道音频信号质量的系统和方法

    公开(公告)号:US07024259B1

    公开(公告)日:2006-04-04

    申请号:US09889697

    申请日:1999-12-15

    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。

    Apparatus and method for suppressing feedback
    38.
    发明申请
    Apparatus and method for suppressing feedback 有权
    用于抑制反馈的装置和方法

    公开(公告)号:US20050190929A1

    公开(公告)日:2005-09-01

    申请号:US11055353

    申请日:2005-02-08

    CPC分类号: H04R3/02

    摘要: An apparatus for suppressing feedback in an environment where a microphone and a loudspeaker are located, comprises a means for embedding a test signal into a loudspeaker signal, a microphone signal or a modified microphone signal, preferably by using a psychoacoustic masking threshold by using a pseudo-noise test signal, a means for determining a characteristic of a transmission channel in the environment between the loudspeaker and the microphone by using the embedded test signal and the microphone signal, a filter for filtering the loudspeaker signal to obtain a filtered loudspeaker signal, wherein the filter is adaptable to be adapted with regard to its filter characteristic to the characteristic of the transmission channel by the means for determining, as well as a means for subtracting the filtered loudspeaker signal from the microphone signal to obtain the modified microphone signal, in which the feedback is reduced due to the loudspeaker signal. The feedback suppression concept provides an effective feedback suppression without audio quality loss, by which particularly an artist is not affected in his artistic performance.

    摘要翻译: 用于抑制麦克风和扬声器所在的环境中的反馈的装置包括用于将测试信号嵌入到扬声器信号,麦克风信号或修改的麦克风信号中的装置,优选地通过使用心理声学屏蔽阈值,通过使用伪 - 噪声测试信号,用于通过使用嵌入的测试信号和麦克风信号来确定扬声器和麦克风之间的环境中的传输信道的特性的装置,用于对扬声器信号进行滤波以获得滤波的扬声器信号的滤波器,其中 该滤波器适于通过用于确定的装置适应于其对传输信道的特性的滤波特性,以及用于从麦克风信号中减去经滤波的扬声器信号以获得修改的麦克风信号的装置,其中 由于扬声器信号,反馈减少。 反馈抑制概念提供了有效的反馈抑制,没有音频质量损失,特别是艺术家在他的艺术表现中不受影响。

    Device and method for analysing a decoded time signal
    39.
    发明申请
    Device and method for analysing a decoded time signal 有权
    用于分析解码时间信号的装置和方法

    公开(公告)号:US20050175252A1

    公开(公告)日:2005-08-11

    申请号:US10220651

    申请日:2001-02-16

    摘要: An apparatus for analyzing an analysis time signal that has been generated from encoding and decoding an original time signal according to an encoding algorithm first, wherein first the encoding block raster underlying the analysis time signal used by the encoding algorithm is determined. Thereupon, the analysis time signal will be converted from its timely representation comprising a plurality of analysis spectral coefficients, to a spectral representation by using the established encoding block raster. Then, at least two analysis spectral coefficients or at least two spectral coefficients derived from the analysis spectral coefficients by multiplication of an encoding amplification factor or by multiplication with a compression function are grouped. Then, the greatest common divisor of the analysis spectral coefficients or the spectral coefficients derived from the analysis spectral coefficients will be calculated, corresponding to the quantization step width used when quantizing the encoding algorithm or an integer multiple of it. Then, in the case of an audio signal, the scale factor can easily be established for this group of spectral coefficients, i.e. for a scale factor band, from the quantization step width. Thus, all parameters used for the quantization of the original time signal are known, so that for quantizing the analysis time signal no longer full iteration loops have to be performed, which are, on the one hand, very computing time intensive and, on the other hand, introduce tandem encoding distortions.

    摘要翻译: 一种用于分析根据编码算法首先对原始时间信号进行编码和解码而产生的分析时间信号的装置,其中首先确定编码算法使用的分析时间信号下面的编码块光栅。 因此,分析时间信号将通过使用所建立的编码块光栅从包括多个分析频谱系数的及时表示转换成频谱表示。 然后,将通过编码放大因子的乘法或通过与压缩函数相乘而从分析频谱系数导出的至少两个分析频谱系数或至少两个频谱系数分组。 然后,对应于当量化编码算法或其整数倍时使用的量化步长,将计算分析频谱系数的最大公约数或从分析频谱系数导出的频谱系数。 然后,在音频信号的情况下,从量化步长可以容易地为该组频谱系数建立比例因子,即缩放因子频带。 因此,用于原始时间信号的量化的所有参数是已知的,使得对于分析时间信号的量化不再必须执行完整的迭代循环,这一方面一方面非常计算时间密集,并且在 另一方面,引入串联编码失真。