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    • 1. 发明申请
    • IMPROVED HARMONIC TRANSPOSITION
    • 改进的谐波传输
    • WO2010086461A1
    • 2010-08-05
    • PCT/EP2010/053222
    • 2010-03-12
    • DOLBY INTERNATIONAL ABEKSTRAND, PerVILLEMOES, Lars, Falck
    • EKSTRAND, PerVILLEMOES, Lars, Falck
    • G10L19/02G10L21/02G10L21/04
    • G10L19/022G10L19/0212G10L19/24G10L21/038G10L21/04
    • The present invention relates to transposing signals in time and/or frequency and in particular to coding of audio signals. More particular, the present invention relates to high frequency reconstruction (HFR) methods including a frequency domain harmonic transposer. A method and system for generating a transposed output signal from an input signal using a transposition factor T is described. The system comprises an analysis window of length L a , extracting a frame of the input signal, and an analysis transformation unit of order M transforming the samples into M complex coefficients. M is a function of the transposition factor T. The system further comprises a nonlinear processing unit altering the phase of the complex coefficients by using the transposition factor T, a synthesis transformation unit of order M transforming the altered coefficients into M altered samples, and a synthesis window of length Ls, generating a frame of the output signal.
    • 本发明涉及在时间和/或频率上转置信号,特别涉及音频信号的编码。 更具体地,本发明涉及包括频域谐波转移器的高频重构(HFR)方法。 描述了使用转置因子T从输入信号生成转置输出信号的方法和系统。 该系统包括长度为La的分析窗口,提取输入信号的帧,以及将样本变换为M个复系数的阶数M的分析变换单元。 M是转置因子T的函数。该系统还包括通过使用转置因子T来改变复系数的相位的非线性处理单元,将改变的系数转换为M个改变的样本的阶数M的合成变换单元,以及 合成窗口长度Ls,产生输出信号的帧。
    • 2. 发明申请
    • ALIASING REDUCTION USING COMPLEX-EXPONENTIAL MODULATED FILTERBANKS
    • 使用复杂调制滤波器减少浪费
    • WO2002080362A1
    • 2002-10-10
    • PCT/SE2002/000626
    • 2002-03-28
    • CODING TECHNOLOGIES SWEDEN ABEKSTRAND, Per
    • EKSTRAND, Per
    • H03H17/00
    • H03H17/0266
    • The present invention proposes a new method and apparatus for the improvement of digital filterbanks, by a complex extension of cosine modulated digital filterbanks. The invention employs complex-exponential modulation of a low-pass prototype filter and a new method for optimizing the characteristics of this filter. The invention substantially reduces artifacts due to aliasing emerging from independent modifications of subband signals, for example when using a filterbank as an spectral equalizer. The invention is preferably implemented in software, running on a standard PC or a digital signal processor (DSP), but can also be hardcoded on a custom chip. The invention offers essential improvements for various types of digital equalizers, adaptive filters, multiband companders and spectral envelope adjusting filterbanks used in high frequency reconstruction (HFR) systems.
    • 本发明提出了一种通过余弦调制数字滤波器组的复数扩展来改进数字滤波器组的新方法和装置。 本发明采用低通原型滤波器的复指数调制和用于优化该滤波器的特性的新方法。 本发明基本上减少了由于子带信号的独立修改引起的混叠所产生的伪像,例如当使用滤波器组作为频谱均衡器时。 本发明优选地以在标准PC或数字信号处理器(DSP)上运行的软件实现,但也可以在定制芯片上被硬编码。 本发明为用于高频重建(HFR)系统的各种类型的数字均衡器,自适应滤波器,多频带压缩器和频谱包络调整滤波器组提供了重要的改进。
    • 4. 发明申请
    • METHOD FOR DETERMINING INVERSE FILTER FROM CRITICALLY BANDED IMPULSE RESPONSE DATA
    • 用于从重要条带式冲突响应数据中确定反向滤波器的方法
    • WO2010120394A3
    • 2011-01-27
    • PCT/US2010020846
    • 2010-01-13
    • DOLBY LAB LICENSING CORPBROWN C PHILLIPEKSTRAND PERSEEFELDT ALAN J
    • BROWN C PHILLIPEKSTRAND PERSEEFELDT ALAN J
    • H04R3/04H04R29/00
    • H04R29/001H04R3/04H04R2430/03
    • A method for determining an inverse filter for altering the frequency response of a loudspeaker so that with the inverse filter applied in the loudspeaker's signal path the inverse-filtered loudspeaker output has a target frequency response, and optionally also applying the inverse filter in the signal path, and a system configured (e.g., a general or special purpose processor programmed and configured) to determine an inverse filter. In some embodiments, the inverse filter corrects the magnitude of the loudspeaker's output. In other embodiments, the inverse filter corrects both the magnitude and phase of the loudspeaker's output. In some embodiments, the inverse filter is determined in the frequency domain by applying eigenfilter theory or minimizing a mean square error expression by solving a linear equation system.
    • 一种用于确定用于改变扬声器的频率响应的逆滤波器的方法,使得利用在扬声器信号路径中应用的反相滤波器,经滤波的扬声器输出具有目标频率响应,并且还可选地还在信号路径中应用逆滤波器 以及配置(例如,编程和配置的通用或专用处理器)以确定逆滤波器的系统。 在一些实施例中,逆滤波器校正扬声器输出的幅度。 在其他实施例中,逆滤波器校正扬声器的输出的幅度和相位。 在一些实施例中,通过应用特征滤波器理论或通过求解线性方程系统来最小化均方误差表达式,在频域中确定逆滤波器。
    • 5. 发明申请
    • EFFICIENT COMBINED HARMONIC TRANSPOSITION
    • 有效的组合谐波传输
    • WO2010136459A1
    • 2010-12-02
    • PCT/EP2010/057176
    • 2010-05-25
    • DOLBY INTERNATIONAL ABEKSTRAND, PerVILLEMOES, LarsHEDELIN, Per
    • EKSTRAND, PerVILLEMOES, LarsHEDELIN, Per
    • G10L21/02G10L21/04
    • G10H1/0091G10H1/125G10H2210/311G10L19/265G10L21/038G10L21/0388
    • The present document relates to audio coding systems which make use of a harmonic transposition method for high frequency reconstruction (HFR), and to digital effect processors, e.g. so-called exciters, where generation of harmonic distortion adds brightness to the processed signal. In particular; a system configured to generate a high frequency component of a signal from a low frequency component of the signal is described, The system may comprise an analysis filter bank (501) configured to provide a set of analysis subband signals from the low frequency component of the signal; wherein the set of analysis subband signals comprises at least two analysis subband signals; wherein the analysis filter bank (501) has a frequency resolution of Δ f , The system further comprises a nonlinear processing unit (502) configured to determine a set of synthesis subband signals from the set of analysis subband signals using a transposition order P ; wherein the set of synthesis subband signals comprises a portion of the set of analysis subband signals phase shifted by an amount derived from the transposition order P ; and a synthesis filter bank (504) configured to generate the high frequency component of the signal from the set of synthesis subband signals; wherein the synthesis filter bank (504) has a frequency resolution of F Δ f ; with F being a resolution factor, with F ≥1; wherein the transposition order P is different from the resolution factor F .
    • 本文件涉及利用用于高频重构(HFR)的谐波转置方法的音频编码系统以及数字效果处理器,例如, 所谓的兴奋剂,其中谐波失真的产生增加处理信号的亮度。 尤其是; 描述了被配置为从信号的低频分量产生信号的高频分量的系统。系统可以包括分析滤波器组(501),其被配置为提供来自所述信号的低频分量的一组分析子带信号 信号; 其中所述一组分析子带信号包括至少两个分析子带信号; 其中所述分析滤波器组(501)具有Δf的频率分辨率。所述系统还包括非线性处理单元(502),其被配置为使用转置顺序P从所述一组分析子带信号中确定一组合成子带信号; 其中所述合成子带信号集合包括所述一组分析子带信号的一部分,所述分析子带信号的相位偏移量是从所述置换顺序P导出的量; 以及合成滤波器组(504),被配置为从所述合成子带信号的集合生成所述信号的高频分量; 其中所述合成滤波器组(504)具有F ff的频率分辨率; F为解析因子,F = 1; 其中转置次数P与分辨率因子F不同
    • 7. 发明申请
    • AUDIO SIGNAL ENCODING OR DECODING
    • 音频信号编码或解码
    • WO2005043511A1
    • 2005-05-12
    • PCT/IB2004/052226
    • 2004-10-28
    • KONINKLIJKE PHILIPS ELECTRONICS N.V.CODING TECHNOLOGIES GMBHVILLEMOES, Lars, F.EKSTRAND, PerPURNHAGEN, HeikoSCHUIJERS, Erik, G., P.DE BONT, Fransiscus M., J.
    • VILLEMOES, Lars, F.EKSTRAND, PerPURNHAGEN, HeikoSCHUIJERS, Erik, G., P.DE BONT, Fransiscus M., J.
    • G10L19/00
    • G10L19/008G10L21/038
    • Encoding an audio signal is provided wherein the audio signal includes a first audio channel and a second audio channel, the encoding comprising subband filtering each of the first audio channel and the second audio channel in a complex modulated filterbank to provide a first plurality of subband signals for the first audio channel and a second plurality of subband signals for the second audio channel, downsampling each of the subband signals to provide a first plurality of downsampled subband signals and a second plurality of downsampled subband signals, further subband filtering at least one of the downsampled subband signals in a further filterbank in order to provide a plurality of sub-subband signals, deriving spatial parameters from the sub-subband signals and from those downsampled subband signals that are not further subband filtered, and deriving a single channel audio signal comprising derived subband signals derived from the first plurality of downsampled subband signals and the second plurality of downsampled subband signals. Further, decoding is provided wherein an encoded audio signal comprising an encoded single channel audio signal and a set of spatial parameters is decoded by decoding the encoded single channel audio channel to obtain a plurality of downsampled subband signals, further subband filtering at least one of the downsampled subband signals in a further filterbank in order to provide a plurality of sub-subband signals, and deriving two audio channels from the spatial parameters, the sub-subband signals and those downsampled subband signals that are not further subband filtered.
    • 提供对音频信号的编码,其中音频信号包括第一音频通道和第二音频通道,该编码包括在复调制滤波器组中对第一音频通道和第二音频通道中的每一个进行子带滤波,以提供第一多个子带信号 对于第一音频通道和用于第二音频通道的第二多个子带信号,对每个子带信号进行下采样以提供第一多个下采样子带信号和第二多个下采样子带信号,进一步子带滤波至少一个 在另一个滤波器组中进行下采样的子带信号,以便提供多个子子带信号,从子子带信号和未进一步子带滤波的那些下采样子带信号导出空间参数,以及导出包括导出的 从第一多个下采样子带信号导出的子带信号和 第二多个下采样子带信号。 此外,提供了解码,其中包括编码的单声道音频信号和一组空间参数的编码音频信号通过对编码的单声道音频声道进行解码以获得多个下采样的子带信号而被解码,进一步的子带滤波至少一个 在另一滤波器组中进行下采样的子带信号,以便提供多个子子带信号,以及从空间参数,子子带信号和未进一步子带滤波的那些下采样子带信号导出两个音频信道。
    • 8. 发明申请
    • LOW DELAY MODULATED FILTER BANK
    • 低延迟调制滤波器
    • WO2010094710A2
    • 2010-08-26
    • PCT/EP2010/051993
    • 2010-02-17
    • DOLBY INTERNATIONAL ABEKSTRAND, Per
    • EKSTRAND, Per
    • H03H17/02
    • H04R3/04G06F17/10G06F17/11G10L19/008G10L19/0204G10L19/0208G10L19/26G10L19/265G10L21/00H03H17/0201H03H17/0248H03H17/0266H03H17/0275H03H17/0294H03H2017/0297H04S7/307
    • The document relates to modulated sub-sampled digital filter banks, as well as to methods and systems for the design of such filter banks. In particular, the present document proposes a method and apparatus for the improvement of low delay modulated digital filter banks. The method employs modulation of an asymmetric low-pass prototype filter and a new method for optimizing the coefficients of this filter. Further, a specific design for a (64) channel filter bank using a prototype filter length of (640) coefficients and a system delay of (319) samples is given. The method substantially reduces artifacts due to aliasing emerging from independent modifications of subband signals, for example when using a filter bank as a spectral equalizer. The method is preferably implemented in software, running on a standard PC or a digital signal processor (DSP), but can also be hardcoded on a custom chip. The method offers improvements for various types of digital equalizers, adaptive filters, multiband companders and spectral envelope adjusting filterbanks used in high frequency reconstruction (HFR) or parametric stereo systems.
    • 该文件涉及调制的次采样数字滤波器组,以及用于设计这种滤波器组的方法和系统。 特别地,本文件提出了一种用于改进低延迟调制数字滤波器组的方法和装置。 该方法采用非对称低通原型滤波器的调制和用于优化滤波器系数的新方法。 此外,给出了使用(640)系数的原型滤波器长度和(319)个样本的系统延迟的(64)信道滤波器组的具体设计。 该方法基本上减少了由于子带信号的独立修改引起的混叠所产生的伪像,例如当使用滤波器组作为频谱均衡器时。 该方法优选地以在标准PC或数字信号处理器(DSP)上运行的软件实现,但也可以在定制芯片上被硬编码。 该方法为各种类型的数字均衡器,自适应滤波器,多频带压缩器以及用于高频重构(HFR)或参数立体声系统的频谱包络调整滤波器组提供了改进。