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    • 5. 发明授权
    • Mixing system for mixing oversampled digital audio signals
    • 用于混合过采样数字音频信号的混音系统
    • US07302303B2
    • 2007-11-27
    • US10502171
    • 2002-12-23
    • Derk ReefmanBernardus Antonius Maria Zwaans
    • Derk ReefmanBernardus Antonius Maria Zwaans
    • G06F17/00
    • G11B20/10527G11B2020/00065
    • A mixing system for mixing a plurality of digital audio signals, at least one of which is a noise-shaped oversampled digital audio signal having a predetermined sampling frequency and bit resolution, said system comprising a summing unit for computing a sum signal of said plurality of input signals; a clipping unit having an input for receiving said sum signal, said clipping unit clipping said sum signal; further comprising a filter unit between the input terminals and the clipping unit, arranged to selectively suppress frequency components outside an audio frequency band from the sum signal; and a converter unit arranged to receive a clipped sum signal from the clipping unit and to convert said clipped sum signal into an output signal of said bit resolution, using noise-shaping, the clipping unit being arranged to limit the input values to a range of values that the converter is able to handle in a stable manner.
    • 一种用于混合多个数字音频信号的混合系统,其中至少一个是具有预定采样频率和比特分辨率的噪声形状的过采样数字音频信号,所述系统包括一个求和单元,用于计算所述多个 输入信号 剪切单元,具有用于接收所述和信号的输入,所述剪切单元限制所述和信号; 进一步包括在所述输入端和所述剪切单元之间的滤波器单元,被布置成从所述和信号有选择地抑制音频频带外的频率分量; 以及转换器单元,其被布置为从所述剪切单元接收剪切和信号,并且使用噪声整形将所述剪切的和信号转换成所述比特分辨率的输出信号,所述剪切单元被布置为将输入值限制到 转换器能够以稳定的方式处理的值。
    • 6. 发明申请
    • Mixing system for mixing oversampled digital audio signals
    • 用于混合过采样数字音频信号的混音系统
    • US20050075744A1
    • 2005-04-07
    • US10502171
    • 2002-12-23
    • Derk ReefmanBernardus Zwaans
    • Derk ReefmanBernardus Zwaans
    • G11B20/00G11B20/10G06F17/00H04B1/00
    • G11B20/10527G11B2020/00065
    • A mixing system for mixing a plurality of digital audio signals, at least one of which is a noise-shaped oversampled digital audio signal having a predetermined sampling frequency and bit resolution, said system comprising a summing unit for computing a sum signal of said plurality of input signals; a clipping unit having an input for receiving said sum signal, said clipping unit clipping said sum signal; further comprising a filter unit between the input terminals and the clipping unit, arranged to selectively suppress frequency components outside an audio frequency band from the sum signal; and a converter unit arranged to receive a clipped sum signal from the clipping unit and to convert said clipped sum signal into an output signal of said bit resolution, using noise-shaping, the clipping unit being arranged to limit the input values to a range of values that the converter is able to handle in a stable manner.
    • 一种用于混合多个数字音频信号的混合系统,其中至少一个是具有预定采样频率和比特分辨率的噪声形状的过采样数字音频信号,所述系统包括一个求和单元,用于计算所述多个 输入信号 剪切单元,具有用于接收所述和信号的输入,所述剪切单元限制所述和信号; 进一步包括在所述输入端和所述剪切单元之间的滤波器单元,被布置成从所述和信号有选择地抑制音频频带外的频率分量; 以及转换器单元,其被布置为从所述剪切单元接收剪切和信号,并且使用噪声整形将所述剪切的和信号转换成所述比特分辨率的输出信号,所述剪切单元被布置为将输入值限制到 转换器能够以稳定的方式处理的值。
    • 9. 发明申请
    • DIRECT STREAM DIGITAL AUDIO WITH MINIMAL STORAGE REQUIREMENT
    • 具有最低存储要求的直播流数字音频
    • US20090287493A1
    • 2009-11-19
    • US11915478
    • 2006-05-16
    • Erwin JanssenDerk Reefman
    • Erwin JanssenDerk Reefman
    • G10L19/00H03M3/00H03M1/06
    • H03M7/3006G10L19/167G11B20/10481G11B20/10527G11B20/1262G11B2020/00014G11B2020/00065G11B2020/10546G11B2020/1288G11B2220/2541G11B2220/2579H03M7/3028
    • An audio coding scheme allowing PCM signal to lossless DSD signal expansion for next generation optical disc formats. The method of encoding an input DSD signal includes up-sampling a corresponding PCM signal to the DSD sample rate. Then a set of loop filter parameters for a noise-shaping loop of a sigma-delta modulator are generated, either using a random starting condition of the sigma-delta modulator or including synchronization parameters. This will allow a decoder to regenerate an almost perfect signal, but still it needs a correction signal to be able to bit identically regenerate the DSD input signal. Therefore, a correction signal is generated based on a difference between a sigma-delta modulated version of the up-sampled PCM signal and the input DSD signal, wherein the sigma-delta modulated version of the up-sampled PCM signal is obtained using the set of loop filter parameters. The correction signal may be adapted to be applied to the low bit PCM signal, to the up-sampled PCM signal or to the output bit stream. Finally, an expansion bit stream is generated where an encoded version of the set of loop filter parameters and an encoded version of the correction signal are included. The decoder can reproduce the original DSD signal based on the already available PCM signal and the described expansion bit stream. Thus, the coding scheme enables top quality audio with minimal storage overhead since the already available PCM signal is used in combination with an expansion bit stream. Since only an additional data stream is required to be stored on a disc, e.g. as part of an MPEG stream, DSD functionality is added to existing systems without causing compatibility problems.
    • 音频编码方案允许PCM信号对下一代光盘格式的无损DSD信号扩展。 编码输入DSD信号的方法包括将对应的PCM信号上采样到DSD采样率。 然后,使用Σ-Δ调制器的随机起始条件或包括同步参数来生成Σ-Δ调制器的噪声整形环路的一组环路滤波器参数。 这将允许解码器重新生成几乎完美的信号,但是仍然需要一个校正信号以能够相同地重新生成DSD输入信号。 因此,基于上采样PCM信号的Σ-Δ调制版本与输入DSD信号之间的差产生校正信号,其中使用该集合获得上采样PCM信号的Σ-Δ调制版本 的环路滤波器参数。 校正信号可适用于低位PCM信号,上采样PCM信号或输出比特流。 最后,生成扩展位流,其中包括一组环路滤波器参数的编码版本和校正信号的编码版本。 解码器可以基于已经可用的PCM信号和所描述的扩展位流再现原始DSD信号。 因此,由于已经可用的PCM信号与扩展位流结合使用,所以编码方案实现了具有最小存储开销的顶级质量音频。 由于只需要另外的数据流来存储在盘上, 作为MPEG流的一部分,DSD功能被添加到现有系统中,而不会引起兼容性问题。