Selecting a coupling scheme for each subband for estimation of coupling parameters in a transform coder for high quality audio
    1.
    发明授权
    Selecting a coupling scheme for each subband for estimation of coupling parameters in a transform coder for high quality audio 有权
    选择每个子带的耦合方案,以估计用于高质量音频的变换编码器中的耦合参数

    公开(公告)号:US06591241B1

    公开(公告)日:2003-07-08

    申请号:US09582552

    申请日:2000-09-08

    CPC classification number: H04S1/007 H04B1/665

    Abstract: A method and apparatus for determining parameters for coupling of channels in a digital audio encoder. A frequency range of two audio channels is coupled together in a coupling channel, and a systematic method of determining optimum coupling parameters is employed. Sub-bands of the channels are processed individually, and a measure of the power of each sub-band is used to determine a coupling coefficient generation scheme for each individual sub-band. Adjacent ones of the sub-bands using the same coupling scheme are combined to form bands in the coupling channel, which dictate the generation of the coupling coordinates for the audio channels. The arrangement of the sub-bands in bands also facilitates the generation of phase flags for each band, on the basis of the coupling scheme used in the band.

    Abstract translation: 一种用于确定数字音频编码器中的信道耦合的参数的方法和装置。 两个音频通道的频率范围在耦合通道中耦合在一起,并且采用确定最佳耦合参数的系统方法。 单独处理信道的子带,并且使用每个子带的功率的度量来确定每个单独子带的耦合系数生成方案。 使用相同耦合方案的相邻的一些子带被组合以在耦合信道中形成频带,这决定了音频信道的耦合坐标的产生。 根据频带中使用的耦合方案,频带中的子频带的布置也有助于为每个频带生成相位标志。

    Method of encoding frequency coefficients in an AC-3 encoder
    2.
    发明授权
    Method of encoding frequency coefficients in an AC-3 encoder 失效
    对AC-3编码器频率系数进行编码的方法

    公开(公告)号:US06775587B1

    公开(公告)日:2004-08-10

    申请号:US10129047

    申请日:2003-01-14

    CPC classification number: G10L19/02

    Abstract: A method for encoding frequency coefficients in an AC-3 Encoder. The method includes: representing frequency coefficients in theform of a respective exponent and mantissa; coding the exponents; and shifting the mantissas to compensate for changes in the exponent values, wherein the exponents comprise an original exponent set (e0, e1, . . . en−1) which is mapped to a new exponent set (e0′, e1′, . . . , e′n−1) after coding, so as to satisfy: ∥e′i+1−e′i∥

    Abstract translation: 一种用于对AC-3编码器中的频率系数进行编码的方法。 该方法包括:表示相应指数和尾数形式中的频率系数; 编码指数; 并且移动尾数以补偿指数值的变化,其中指数包括被映射到新的指数集合(e0',e1',...)的原始指数集合(e0,e1,...,en-1)。 ,e'n-1),以满足:|| e'i + 1-e'i || D,其中i = 0。 。 。 ,n-1和D是两个连续指数之间的最大允许差,e'i <= Ei。

    Dual channel phase flag determination for coupling bands in a transform coder for high quality audio
    3.
    发明授权
    Dual channel phase flag determination for coupling bands in a transform coder for high quality audio 有权
    在用于高质量音频的变换编码器中耦合频带的双通道相位标志确定

    公开(公告)号:US06574602B1

    公开(公告)日:2003-06-03

    申请号:US09581779

    申请日:2000-09-08

    CPC classification number: H04B1/665 H04S1/007

    Abstract: A method and apparatus for subband phase flag determination for coupling of channels in a dual channel audio encoder is based on a psychoacoustic model of the human auditory system. The method and apparatus are applicable to audio encoders which utilize a coupling channel to combine certain frequency components of the input audio signals. The method ensures a least square error between the original channel frequency coefficients at the encoder and the estimated coefficients at the decoder by determining the sign of the dot product of the coefficients for one of the channels and the coupling coefficients. No restriction is placed on the strategy utilized for generating the coupling channel coefficients or the coupling coordinates.

    Abstract translation: 用于双通道音频编码器中的信道耦合的子带相位标志确定的方法和装置基于人类听觉系统的心理声学模型。 该方法和装置适用于利用耦合通道组合输入音频信号的某些频率分量的音频编码器。 该方法通过确定通道之一和耦合系数的系数的积积的符号来确保编码器处的原始通道频率系数和解码器处的估计系数之间的最小平方误差。 对用于产生耦合通道系数或耦合坐标的策略没有限制。

    Non-uniform filter bank implementation
    4.
    发明授权
    Non-uniform filter bank implementation 失效
    非均匀滤波器组实现

    公开(公告)号:US07424502B2

    公开(公告)日:2008-09-09

    申请号:US10490727

    申请日:2001-09-28

    CPC classification number: H03H17/0266

    Abstract: A method of searching for a best-match decimation vector of decimation factors for non-uniform filter bank, the best match vector allowing perfect or near-perfect reconstruction of an input signal of the non-uniform filter bank, the method including the steps of: a) selecting a partial decimation vector having a number, l, of decimation factors, where l does not exceed a maximum number, K, of decimation factors of said best-match decimation vector; b) testing said l decimation factors to determine whether said partial decimation vector satisfies a feasibility criterion; c) testing a least common multiplier value of said l decimation factors to determine whether said least common multiplier value is greater than a predetermined value; d) testing a maximum decimation value, Dmax, of said partial decimation vector to determine whether Dmax is less than one; e) testing a minimum decimation value, Dmin, of said partial decimation vector to determine whether Dmin is greater than one; and f) if said feasibility criterion is satisfied and Dmax is not less than one and Dmin is not greater than one, then incrementing by one the number l of decimation factors in the partial decimation vector and repeating steps b) to e) for a plurality of times.

    Abstract translation: 一种搜索用于非均匀滤波器组的抽取因子的最佳匹配抽取向量的方法,所述最佳匹配向量允许非均匀滤波器组的输入信号的完美或接近完美的重建,所述方法包括以下步骤: :a)选择具有数字l抽取因子的部分抽取向量,其中l不超过所述最佳匹配抽取向量的抽取因子的最大数目K; b)测试所述l个抽取因子以确定所述部分抽取向量是否满足可行性标准; c)测试所述l个抽取因子的最小公式倍数值,以确定所述最小公倍数值是否大于预定值; d)测试所述部分抽取向量的最大抽取值D max max,以确定D MAX max是否小于1; e)测试所述部分抽取向量的最小抽取值D min min以确定D min min是否大于1; 以及f)如果满足所述可行性标准,并且D最大值不小于1,并且D分钟不大于1,则将数字l的抽取因子加1 在部分抽取向量中重复步骤b)至e)多次。

    Method of encoding an audio signal using a quality value for bit allocation
    6.
    发明授权
    Method of encoding an audio signal using a quality value for bit allocation 有权
    使用用于比特分配的质量值对音频信号进行编码的方法

    公开(公告)号:US07003449B1

    公开(公告)日:2006-02-21

    申请号:US10129045

    申请日:1999-10-30

    CPC classification number: G10L19/0208 G10L19/002 G10L19/035

    Abstract: A method for encoding an audio signal, including providing a masking function, representative of psychoacoustic masking; setting a quality value for data of the encoded signal, adjusting the masking function dependent upon the quality value; and allocating bits for quantization of the encoded signal based on the incremental masking function.

    Abstract translation: 一种用于编码音频信号的方法,包括提供表示心理声学掩蔽的掩蔽功能; 为编码信号的数据设置质量值,根据质量值调整掩蔽函数; 以及基于增量掩蔽功能分配用于量化编码信号的比特。

    Method and apparatus for spectral exponent reshaping in a transform coder for high quality audio
    7.
    发明授权
    Method and apparatus for spectral exponent reshaping in a transform coder for high quality audio 有权
    用于高质量音频的变换编码器中的频谱指数整形的方法和装置

    公开(公告)号:US06839674B1

    公开(公告)日:2005-01-04

    申请号:US09582766

    申请日:1998-01-12

    CPC classification number: H03M7/3002 H04B1/665

    Abstract: A method and apparatus for coding audio data in a frequency transform digital audio coder employing differential frequency coefficient exponent coding. Differential coding of exponents places constraints on possible values an exponent can take, which can lead to distortion in the decoded and reconstructed audio signal. The method and apparatus herein can overcome this restriction by mapping the input exponent set to a new set of values which satisfy the differential constraint as well as reducing information loss, thereby minimizing overall signal distortion due to coding restrictions.

    Abstract translation: 一种用于使用差分频率系数指数编码的频率变换数字音频编码器中的音频数据编码的方法和装置。 指数的差分编码对指数可以采用的可能值设置约束,这可能导致解码和重构的音频信号中的失真。 这里的方法和装置可以通过将输入指数集合映射到满足差分约束的新的一组值以及减少信息损失来克服这种限制,从而最小化由编码限制引起的整体信号失真。

    Loop Invariant Method Expression Hoisting
    8.
    发明申请
    Loop Invariant Method Expression Hoisting 有权
    循环不变方法表达式起重

    公开(公告)号:US20140173576A1

    公开(公告)日:2014-06-19

    申请号:US13713521

    申请日:2012-12-13

    CPC classification number: G06F8/4441 G06F8/37

    Abstract: A system, method and computer-readable medium are disclosed for improving the performance of a compiler. A set of source code instructions are processed to generate a plurality of source code instruction subsets, each of which is respectively associated with a mathematical operator. The source code subsets are then reordered to “hoist,” or place, a source code instruction subset associated with a product operator before a source code instruction subset associated with a summation operator. The plurality of source code instruction subsets are iteratively reordered until no source code instruction subset associated with a summation operator precedes a source code instruction subset associated with a product operator. A compiler is then used to compile the resulting reordered plurality of source code instruction subsets into a set of optimized object code instructions.

    Abstract translation: 公开了一种用于提高编译器的性能的系统,方法和计算机可读介质。 处理一组源代码指令以生成多个源代码指令子集,每个源代码指令子集分别与数学运算符相关联。 然后将源代码子集重新排序为“与起始运算符相关联的源代码指令子集”之前的“提升”或放置与产品运算符相关联的源代码指令子集。 多个源代码指令子集被迭代地重新排序,直到与求和运算符相关联的源代码指令子集在与产品运算符相关联的源代码指令子集之前。 然后使用编译器将所得到的重新排序的多个源代码指令子集编译成一组优化的目标代码指令。

    Unified filter bank for audio coding
    9.
    发明授权
    Unified filter bank for audio coding 有权
    用于音频编码的统一滤波器组

    公开(公告)号:US07369989B2

    公开(公告)日:2008-05-06

    申请号:US10479952

    申请日:2001-06-08

    CPC classification number: G10L19/0208

    Abstract: A unified filter bank for use in encoding and decoding MPEG-1 audio data, wherein input audio data is encoded into coded audio data and the coded audio data is subsequently decoded into output audio data. The unified filter bank includes a plurality of filters, with each filter of the plurality of filters being a cosine modulation of a prototype filter. The unified filter bank is operational as an analysis filter bank during audio data encoding and as a synthesis filter bank during audio data decoding, wherein the unified filter bank is effective to substantially eliminate the effects of aliasing, phase distortion and amplitude distortion in the output audio data.

    Abstract translation: 用于对MPEG-1音频数据进行编码和解码的统一滤波器组,其中将输入音频数据编码为经编码的音频数据,随后将经编码的音频数据解码为输出音频数据。 统一滤波器组包括多个滤波器,多个滤波器的每个滤波器是原型滤波器的余弦调制。 在音频数据编码期间,统一的滤波器组可用作分析滤波器组,并且在音频数据解码期间用作合成滤波器组,其中统一滤波器组有效地基本上消除输出音频中的混叠,相位失真和幅度失真的影响 数据。

    Method and system for parametric characterization of transient audio signals
    10.
    发明授权
    Method and system for parametric characterization of transient audio signals 有权
    瞬态音频信号的参数化表征方法和系统

    公开(公告)号:US07363216B2

    公开(公告)日:2008-04-22

    申请号:US10626845

    申请日:2003-07-23

    CPC classification number: G10L19/025

    Abstract: A method of parametrically encoding a transient audio signal, including the steps of: determining a set V of the N largest frequency components of the transient audio signal, where N is a predetermined number; determining an approximate envelope of the transient audio signal; and determining a predetermined number P of samples W of the approximate envelope for use in generating a spline approximation of the approximate envelope, whereby a parametric representation of the transient audio signal is given by parameters including V, N, P and W, such that a decoder receiving the parametric representation can reproduce a received approximation of the transient audio signal.

    Abstract translation: 一种对瞬态音频信号进行参数编码的方法,包括以下步骤:确定瞬态音频信号的N个最大频率分量的集合V,其中N是预定数目; 确定瞬态音频信号的近似包络; 以及确定用于产生近似包络的样条近似的近似包络的预定数量P的样本W,由此通过包括V,N,P和W的参数给出瞬态音频信号的参数表示,使得 接收参数表示的解码器可以再现接收到的瞬态音频信号的近似值。

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