Equalization for DMT and OFDM communication systems
    1.
    发明授权
    Equalization for DMT and OFDM communication systems 有权
    DMT和OFDM通信系统的均衡

    公开(公告)号:US07433398B2

    公开(公告)日:2008-10-07

    申请号:US10889530

    申请日:2004-07-12

    IPC分类号: H03K5/159

    摘要: The invention provides a DMT or OFDM equalization system that deals with each tone not only separately, but also globally, which provides better overall performance. In the inventive system, only M+T variables are required to be determined, where M is the number of tones and T represents the number of SIRF taps. The invention defines a matrix R, in which optimal coefficients are found as the eigenvector corresponding to the smallest eigenvalue of the matrix R. The dynamic range of all variables is limited in a system in accordance with the present invention, which provides ease of hardware implementation. Furthermore, the inventive system retains the advantages of per-tone equalization by providing a smoother signal-to-noise ratio (SNR) distribution function versus synchronization delay. In addition, no effort is wasted on the equalization of unused tones, because it is unnecessary to determine the coefficients for unused tones.

    摘要翻译: 本发明提供了一种DMT或OFDM均衡系统,其不仅分别地,而且全局地处理每个音调,这提供了更好的整体性能。 在本发明的系统中,仅需要确定M + T个变量,其中M是音调的数量,T表示SIRF抽头的数量。 本发明定义了矩阵R,其中找到最佳系数作为与矩阵R的最小特征值对应的特征向量。所有变量的动态范围在根据本发明的系统中受到限制,这提供了硬件实现的容易性 。 此外,本发明的系统通过相对于同步延迟提供更平滑的信噪比(SNR)分布函数来保留每个音调均衡的优点。 此外,由于不需要确定未使用音调的系数,因此不会浪费对未使用音调的均衡。

    Equalization for DMT and OFDM communication systems
    2.
    发明申请
    Equalization for DMT and OFDM communication systems 有权
    DMT和OFDM通信系统的均衡

    公开(公告)号:US20060007998A1

    公开(公告)日:2006-01-12

    申请号:US10889530

    申请日:2004-07-12

    IPC分类号: H03H7/30 H04K1/10

    摘要: The invention provides a DMT or OFDM equalization system that deals with each tone not only separately, but also globally, which provides better overall performance. In the inventive system, only M+T variables are required to be determined, where M is the number of tones and T represents the number of SIRF taps. The invention defines a matrix R, in which optimal coefficients are found as the eigenvector corresponding to the smallest eigenvalue of the matrix R. The dynamic range of all variables is limited in a system in accordance with the present invention, which provides ease of hardware implementation. Furthermore, the inventive system retains the advantages of per-tone equalization by providing a smoother signal-to-noise ratio (SNR) distribution function versus synchronization delay. In addition, no effort is wasted on the equalization of unused tones, because it is unnecessary to determine the coefficients for unused tones.

    摘要翻译: 本发明提供了一种DMT或OFDM均衡系统,其不仅分别地,而且全局地处理每个音调,这提供了更好的整体性能。 在本发明的系统中,仅需要确定M + T个变量,其中M是音调的数量,T表示SIRF抽头的数量。 本发明定义了矩阵R,其中找到最佳系数作为与矩阵R的最小特征值对应的特征向量。所有变量的动态范围在根据本发明的系统中受到限制,这提供了硬件实现的容易性 。 此外,本发明的系统通过相对于同步延迟提供更平滑的信噪比(SNR)分布函数来保留每个音调均衡的优点。 此外,由于不需要确定未使用音调的系数,因此不会浪费对未使用音调的均衡。

    Method and apparatus for an adaptive binaural beamforming system

    公开(公告)号:US06983055B2

    公开(公告)日:2006-01-03

    申请号:US10006086

    申请日:2001-12-05

    申请人: Fa-Long Luo

    发明人: Fa-Long Luo

    IPC分类号: H04R25/00

    摘要: An adaptive binaural beamforming system is provided which can be used, for example, in a hearing aid. The system uses more than two input signals, and preferably four input signals. The signals can be provided, for example, by two microphone pairs, one pair of microphones located in a user's left ear and the second pair of microphones located in the user's right ear. The system is preferably arranged such that each pair of microphones utilizes an end-fire configuration with the two pairs of microphones being combined in a broadside configuration. Signal processing is divided into two stages. In the first stage, the outputs from the two microphone pairs are processed utilizing an end-fire array processing scheme, this stage providing the benefits of spatial processing. In the second stage, the outputs from the two end-fire arrays are processed utilizing a broadside configuration, this stage providing further spatial processing benefits along with the benefits of binaural processing.

    IC for universal computing with near zero programming complexity
    4.
    发明授权
    IC for universal computing with near zero programming complexity 有权
    用于具有接近零编程复杂度的通用计算的IC

    公开(公告)号:US06947916B2

    公开(公告)日:2005-09-20

    申请号:US10029502

    申请日:2001-12-21

    IPC分类号: G06N3/063 G06F15/18

    CPC分类号: G06N3/063

    摘要: A computing machine capable of performing multiple operations using a universal computing unit is provided. The universal computing unit maps an input signal to an output signal. The mapping is initiated using an instruction that includes the input signal, a weight matrix, and an activation function. Using the instruction, the universal computing unit may perform multiple operations using the same hardware configuration. The computation that is performed by the universal computing unit is determined by the weight matrix and activation function used. Accordingly, the universal computing unit does not require any programming to perform a type of computing operation because the type of operation is determined by the parameters of the instruction, specifically, the weight matrix and the activation function.

    摘要翻译: 提供了能够使用通用计算单元执行多个操作的计算机。 通用计算单元将输入信号映射到输出信号。 使用包括输入信号,权重矩阵和激活功能的指令来启动映射。 使用该指令,通用计算单元可以使用相同的硬件配置来执行多个操作。 由通用计算单元执行的计算由所使用的权重矩阵和激活函数确定。 因此,通用计算单元不需要任何编程来执行一种计算操作,因为操作的类型由指令的参数,具体地,权重矩阵和激活功能确定。

    Method and system for providing stereo-channel based multi-channel audio coding
    5.
    发明授权
    Method and system for providing stereo-channel based multi-channel audio coding 有权
    用于提供基于立体声道的多声道音频编码的方法和系统

    公开(公告)号:US08041041B1

    公开(公告)日:2011-10-18

    申请号:US11443878

    申请日:2006-05-30

    IPC分类号: H04R5/00

    摘要: A system for generating stereo-channel audio signals with surround information is disclosed. The system includes a surround mapping unit configured to receive signals from a number of audio channels and generate a pair of stereo-channel audio signals based on the audio channels. The pair of stereo-channel audio signals includes binaural and spatial information. The system also includes a stereo-channel encoder configured to receive and encode the pair of stereo-channel audio signals from the surround mapping unit thereby generating a pair of encoded stereo-channel audio signals. The system further includes a stereo-channel decoder configured to receive and decode the pair of encoded stereo-channel audio signals thereby obtaining the pair of stereo-channel audio signals. The pair of stereo-channel audio signals are capable of being used to generate surround effect.

    摘要翻译: 公开了一种用于产生具有环绕信息的立体声道音频信号的系统。 该系统包括环绕映射单元,其被配置为从多个音频通道接收信号,并且基于音频通道生成一对立体声声道音频信号。 该对立体声道音频信号包括双耳和空间信息。 该系统还包括立体声通道编码器,其被配置为从环绕映射单元接收和编码一对立体声声道音频信号,从而生成一对经编码的立体声道音频信号。 该系统还包括立体声声道解码器,其被配置为接收和解码该对编码的立体声声道音频信号,从而获得一对立体声声道音频信号。 一对立体声声道音频信号能够用于产生环绕声效果。

    Method and system for implementing a low-complexity scheme in color conversion and down-sampling of image codecs
    6.
    发明授权
    Method and system for implementing a low-complexity scheme in color conversion and down-sampling of image codecs 失效
    在图像编解码器的颜色转换和下采样中实现低复杂度方案的方法和系统

    公开(公告)号:US07693326B1

    公开(公告)日:2010-04-06

    申请号:US10263941

    申请日:2002-10-02

    申请人: Fa-Long Luo

    发明人: Fa-Long Luo

    IPC分类号: G06K9/00 G06K9/32

    CPC分类号: H04N19/186 H04N19/132

    摘要: An image processing method and system using a low-complexity scheme is provided. According to one aspect of the method, input components from a RGB model are used directly to calculate the down-sampled components of a YCbCr model. In an exemplary instance where average down-sampling and a down-sampling rate of “4:2:0” are used, the following equations are used to derive the down-sampled components of the YCbCr model: Y l ″ = ⁢ 0.29900 ⁢ R i + 0.58700 ⁢ G l + 0.11400 ⁢ B l Cb ″ = ⁢ - 0.04219 ⁢ ∑ i = 0 3 ⁢ ⁢ R l - 0.082815 ⁢ ∑ i = 0 3 ⁢ ⁢ G l + 0.12500 ⁢ ∑ i = 0 3 ⁢ ⁢ B l + 2 SP / 2 Cr ″ = ⁢ 0.12500 ⁢ ∑ i = 0 3 ⁢ ⁢ R i - 0.10467 ⁢ ∑ i = 0 3 ⁢ ⁢ G i - 0.02033 ⁢ ∑ i = 0 3 ⁢ ⁢ B i + 2 SP / 2 where Ri, Gi and Bi are three input components of the color conversion for the pixel i and “SP” represents a specified sample precision under the RGB model. The foregoing method reduces computational complexity and cost thereby allowing an image color conversion process to be performed in a more efficient manner.

    摘要翻译: 提供了一种使用低复杂度方案的图像处理方法和系统。 根据该方法的一个方面,来自RGB模型的输入分量被直接用于计算YCbCr模型的下采样分量。 在使用平均下采样和下采样率“4:2:0”的示例性实例中,使用以下等式来导出YCbCr模型的下采样分量:Y l“= 0.29900 R i + 0.58700 G l + 0.11400 B l Cb“= - 0.04219Σi = 0 3·保守R 1-0.082815Σi= 0 3·塞尔+ 1 + B l + 2 SP / 2 Cr“=0.12500Σi= 0 3·保守R i-0.10467Σi= 0 3·塞尔G i - 0.02033Σi = 0 3⁢B i + 2 SP / 2其中Ri,Gi和Bi是像素i的颜色转换的三个输入分量,“SP”表示RGB模型下的指定采样精度。 上述方法降低了计算复杂性和成本,从而允许以更有效的方式执行图像颜色转换处理。

    Method and system for providing an excitation-pattern based audio coding scheme
    7.
    发明授权
    Method and system for providing an excitation-pattern based audio coding scheme 有权
    用于提供基于激励模式的音频编码方案的方法和系统

    公开(公告)号:US07617100B1

    公开(公告)日:2009-11-10

    申请号:US10340060

    申请日:2003-01-10

    申请人: Fa-Long Luo

    发明人: Fa-Long Luo

    IPC分类号: G10L19/00

    CPC分类号: G10L19/032

    摘要: An improved audio compression scheme is provided. The scheme uses an excitation pattern to more efficiently provide audio signal compression. Under the scheme, an input signal is transformed to the frequency domain. Next, the excitation pattern corresponding to the transformed input signal is calculated. Bit allocation processing is then performed based on the excitation pattern. Frequencies are then coded based on the results of the bit allocation processing. Finally, bitstream packing is performed to generate the output coded audio bit stream. In one exemplary implementation, the audio compression scheme is implemented in an encoder.

    摘要翻译: 提供了改进的音频压缩方案。 该方案使用激励模式更有效地提供音频信号压缩。 在该方案下,输入信号被转换到频域。 接下来,计算与变换后的输入信号对应的激励图案。 然后基于激励模式进行位分配处理。 然后根据比特分配处理的结果对频率进行编码。 最后,执行比特流打包以产生输出编码音频比特流。 在一个示例性实现中,音频压缩方案在编码器中实现。

    Two-stage adaptive feedback cancellation scheme for hearing instruments
    8.
    发明授权
    Two-stage adaptive feedback cancellation scheme for hearing instruments 失效
    用于听力仪器的两级自适应反馈取消方案

    公开(公告)号:US06754356B1

    公开(公告)日:2004-06-22

    申请号:US09684294

    申请日:2000-10-06

    IPC分类号: H04R2500

    CPC分类号: H04R25/453

    摘要: A feedback cancellation system for a hearing instrument is described. The feedback cancellation system includes both a constrained adaptive filter, the constrained adaptive filter can be constrained by initiation coefficients, as well as an adaptive gain modification. The adaptive gain modification allows the feedback cancellation to respond to substantially different environments from the test environment used to derive the initiation coefficients for the constrained adaptive filter.

    摘要翻译: 描述了用于听力仪器的反馈消除系统。 反馈消除系统包括约束自适应滤波器,约束自适应滤波器可以由起始系数以及自适应增益修改来约束。 自适应增益修改允许反馈消除响应来自用于导出约束自适应滤波器的起始系数的测试环境的实质上不同的环境。

    System for frequency-domain scaling for discrete cosine transform
    9.
    发明授权
    System for frequency-domain scaling for discrete cosine transform 有权
    用于离散余弦变换的频域缩放系统

    公开(公告)号:US07330866B2

    公开(公告)日:2008-02-12

    申请号:US10612202

    申请日:2003-07-01

    申请人: Fa-Long Luo

    发明人: Fa-Long Luo

    IPC分类号: G06F17/14

    CPC分类号: G06F17/147

    摘要: A system for frequency-domain scaling for DCT computation. Scale factors are applied to coefficients during the final steps of composition of 2-point DCTs. The number of multiplications and required precision are reduced. Fixed values for various scale factors can be computed and stored prior to executing the DCT so that performance can be improved. The fixed values are derived by knowing the length of the time-domain sequence. Some fixed values can be derived independently of the length of the time-domain sequence. The approach of the invention can also reduce the number of multiplications to compute the transform, and allow smaller bit-width sizes by reducing the number of required high-precision calculations.

    摘要翻译: 用于DCT计算的频域缩放系统。 在2点DCT的组成的最后步骤中,将比例因子应用于系数。 减少乘法次数和所需精度。 可以在执行DCT之前计算和存储各种比例因子的固定值,从而可以提高性能。 通过知道时域序列的长度导出固定值。 可以独立于时域序列的长度导出一些固定值。 本发明的方法还可以减少用于计算变换的乘法次数,并且通过减少所需的高精度计算的数量来允许较小的位宽度大小。

    System for frequency-domain scaling for discrete cosine transform
    10.
    发明申请
    System for frequency-domain scaling for discrete cosine transform 有权
    用于离散余弦变换的频域缩放系统

    公开(公告)号:US20050004964A1

    公开(公告)日:2005-01-06

    申请号:US10612202

    申请日:2003-07-01

    申请人: Fa-Long Luo

    发明人: Fa-Long Luo

    IPC分类号: G06F17/14

    CPC分类号: G06F17/147

    摘要: A system for frequency-domain scaling for DCT computation. Scale factors are applied to coefficients during the final steps of composition of 2-point DCTs. The number of multiplications and required precision are reduced. Fixed values for various scale factors can be computed and stored prior to executing the DCT so that performance can be improved. The fixed values are derived by knowing the length of the time-domain sequence. Some fixed values can be derived independently of the length of the time-domain sequence. The approach of the invention can also reduce the number of multiplications to compute the transform, and allow smaller bit-width sizes by reducing the number of required high-precision calculations.

    摘要翻译: 用于DCT计算的频域缩放系统。 在2点DCT的组成的最后步骤中,将比例因子应用于系数。 减少乘法次数和所需精度。 可以在执行DCT之前计算和存储各种比例因子的固定值,从而可以提高性能。 通过知道时域序列的长度导出固定值。 可以独立于时域序列的长度导出一些固定值。 本发明的方法还可以减少用于计算变换的乘法次数,并且通过减少所需的高精度计算的数量来允许较小的位宽度大小。