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    • 12. 发明授权
    • Multiple description transform coding of audio using optimal transforms of arbitrary dimension
    • 使用任意维度的最优变换对音频进行多重变换编码
    • US06253185B1
    • 2001-06-26
    • US09190908
    • 1998-11-12
    • Ramon AreanVivek K. GoyalJelena Kovacevic
    • Ramon AreanVivek K. GoyalJelena Kovacevic
    • G01L1900
    • H04S1/00
    • A multiple description (MD) joint source-channel (JSC) encoder in accordance with the invention encodes n components of an audio signal for transmission over m channels of a communication medium, where n and m may take on any desired values. In an illustrative embodiment, the encoder combines a multiple description transform coder with elements of a perceptual audio coder (PAC). The encoder is configured to select one or more transform parameters for a multiple description transform, based on a characteristic of the audio signal to be encoded. For example, the transform parameters may be selected such that the resulting transformed coefficients have a variance distribution of a type expected by a subsequent entropy coding operation. The components of the audio signal may be quantized coefficients separated into a number of factor bands, and the transform parameter for a given factor band may be set to a value determined based on a transform parameter from at least one other factor band, e.g., the previous factor band. As another example, the transform parameter for one or more of the factor bands may be selected based on a determination as to whether the audio signal to be encoded is of a particular predetermined type. A desired variance distribution may also be obtained for the transformed coefficients by, e.g., pairing or otherwise grouping coefficients such that the coefficients of each pair or group are required to be in the same factor band.
    • 根据本发明的多描述(MD)联合源信道(JSC)编码器编码音频信号的n个分量,用于在通信介质的m个信道上进行传输,其中n和m可以采用任何期望的值。 在说明性实施例中,编码器将多重描述变换编码器与感知音频编码器(PAC)的元件组合。 编码器被配置为基于要编码的音频信号的特性来选择用于多描述变换的一个或多个变换参数。 例如,可以选择变换参数,使得所得到的经变换的系数具有由后续熵编码操作所期望的类型的方差分布。 音频信号的分量可以是分离成多个因子频带的量化系数,并且给定因子频带的变换参数可以被设置为基于来自至少一个其他因子频带的变换参数确定的值,例如, 以前的因子带。 作为另一示例,可以基于关于要编码的音频信号是否是特定预定类型的确定来选择一个或多个因子频带的变换参数。 也可以通过例如配对或以其他方式对系数进行分组来获得变换系数的期望方差分布,使得每对或组的系数被要求处于相同的因子带中。
    • 13. 发明授权
    • Energy-efficient time-stampless adaptive nonuniform sampling
    • 节能时间自适应非均匀采样
    • US09294113B2
    • 2016-03-22
    • US13542070
    • 2012-07-05
    • Soheil Feizi-KhankandiVivek K. GoyalMuriel Médard
    • Soheil Feizi-KhankandiVivek K. GoyalMuriel Médard
    • H03M1/12
    • H03M1/1265
    • Described herein is a sampling system and related sampling scheme. The system and sampling scheme is based upon a framework for adaptive non-uniform sampling schemes. In the system and schemes described herein, time intervals between samples can be computed by using a function of previously taken samples. Therefore, keeping sampling times (time-stamps), except initialization times, is not necessary. One aim of this sampling framework is to provide a balance between reconstruction distortion and average sampling rate. The function by which sampling time intervals can be computed is called the sampling function. The sampling scheme described herein can be applied appropriately on different signal models such as deterministic or stochastic, and continuous or discrete signals. For each different signal model, sampling functions can be derived.
    • 这里描述的是采样系统和相关采样方案。 系统和采样方案基于自适应非均匀采样方案的框架。 在本文描述的系统和方案中,样本之间的时间间隔可以通过使用先前采集的样本的函数来计算。 因此,除了初始化时间之外,不需要保持采样时间(时间戳)。 这个抽样框架的一个目标是提供重建失真和平均采样率之间的平衡。 可以计算采样时间间隔的功能称为采样功能。 本文描述的采样方案可以适当地应用于不同的信号模型,例如确定性或随机的以及连续或离散的信号。 对于每个不同的信号模型,可以导出采样函数。
    • 15. 发明申请
    • Energy-Efficient Time-Stampless Adaptive Nonuniform Sampling
    • 节能时间自适应不均匀采样
    • US20140184273A1
    • 2014-07-03
    • US13542070
    • 2012-07-05
    • Soheil Feizi-KhankandiVivek K. GoyalMuriel Médard
    • Soheil Feizi-KhankandiVivek K. GoyalMuriel Médard
    • H03M1/12
    • H03M1/1265
    • Described herein is a sampling system and related sampling scheme. The system and sampling scheme is based upon a framework for adaptive non-uniform sampling schemes. In the system and schemes described herein, time intervals between samples can be computed by using a function of previously taken samples. Therefore, keeping sampling times (time-stamps), except initialization times, is not necessary. One aim of this sampling framework is to provide a balance between reconstruction distortion and average sampling rate. The function by which sampling time intervals can be computed is called the sampling function. The sampling scheme described herein can be applied appropriately on different signal models such as deterministic or stochastic, and continuous or discrete signals. For each different signal model, sampling functions can be derived.
    • 这里描述的是采样系统和相关采样方案。 系统和采样方案基于自适应非均匀采样方案的框架。 在本文描述的系统和方案中,样本之间的时间间隔可以通过使用先前采集的样本的函数来计算。 因此,除了初始化时间之外,不需要保持采样时间(时间戳)。 这个抽样框架的一个目标是提供重建失真和平均采样率之间的平衡。 可以计算采样时间间隔的功能称为采样功能。 本文描述的采样方案可以适当地应用于不同的信号模型,例如确定性或随机的以及连续或离散的信号。 对于每个不同的信号模型,可以导出采样函数。
    • 16. 发明授权
    • Methods and apparatus for adaptive signal processing involving a Karhunen-Loève basis
    • 涉及Karhunen-Loève基础的自适应信号处理的方法和装置
    • US06993477B1
    • 2006-01-31
    • US09590251
    • 2000-06-08
    • Vivek K. Goyal
    • Vivek K. Goyal
    • G10L21/06
    • G06K9/6247G06F17/14G10L25/48
    • A signal processing device utilizes a stochastic approximation of a gradient descent algorithm for updating a transform. The signal processing device is configured to implement the transform for producing a desired transformed output signal, and the transform is updated using the stochastic approximation of the gradient algorithm based on received data associated with the signal being processed. The transform is represented in a reduced-parameter form, such as a Givens parameterized form or a Householder form, such that the reduced-parameter form for an N×N transform comprises fewer than N2 parameters. The updating process is implemented using computations involving the reduced-parameter form, and an adaptation of the transform is represented directly as one or more changes in the reduced-parameter form. The gradient algorithm may be configured to minimize a negative gradient of a pairwise energy compaction property of the transform. Advantageously, the gradient algorithm may be made locally convergent in mean for a specified step size. The invention can also be implemented in a backward adaptive form in which the updating process is driven by quantized data.
    • 信号处理装置利用梯度下降算法的随机近似来更新变换。 信号处理装置被配置为实现用于产生期望的变换输出信号的变换,并且使用基于与正在处理的信号相关联的接收数据的梯度算法的随机近似来更新变换。 变换以诸如Givens参数化形式或Householder形式的缩减参数形式表示,使得N×N变换的缩减参数形式包括少于N 2 N 2个参数。 使用涉及缩减参数形式的计算来实现更新过程,并且将变换的适应直接表示为缩减参数形式中的一个或多个变化。 梯度算法可以被配置为使变换的成对能量压缩属性的负梯度最小化。 有利地,梯度算法可以对于指定的步长进行局部收敛。 本发明还可以以后向自适应形式实现,其中更新过程由量化数据驱动。
    • 17. 发明授权
    • Methods and apparatus for lattice-structured multiple description vector quantization coding
    • 网格结构多重描述矢量量化编码方法与装置
    • US06594627B1
    • 2003-07-15
    • US09533232
    • 2000-03-23
    • Vivek K. GoyalJonathan Adam KelnerJelena Kovacevic
    • Vivek K. GoyalJonathan Adam KelnerJelena Kovacevic
    • G10L1912
    • G10L19/12G10L2019/0004
    • A lattice-structured multiple description vector quantization (LSMDVQ) encoder generates M descriptions of a signal to be encoded, each of the descriptions being transmittable over a corresponding one of M channels. The encoder is configured based at least in part on a distortion measure which is a function of a central distortion and at least one side distortion. For example, if M=2, the distortion measure may be an average mean-squared error (AMSE) function of the form ƒ(D0, D1, D2), where D0 is a central distortion resulting from reconstruction based on receipt of both a first and a second description, and D1 and D2 are side distortions resulting from reconstruction using only a first description and a second description, respectively. Further performance improvements may be obtained through perturbation of the lattice points. The LSMDVQ techniques of the invention can also be extended to cases of M greater than two, for which the encoder may utilize an ordered set of M codebooks &Lgr;1, &Lgr;2, . . . , &Lgr;M of increasing size, with the coarsest codebook corresponding to a lattice. In such cases, for each number k of descriptions received, there may be a single decoding function that maps the received vector to a corresponding one of the codebooks &Lgr;k, such that reconstruction of the signal requires no more than M such decoding functions.
    • 格子结构的多描述矢量量化(LSMDVQ)编码器生成要编码的信号的M个描述,每个描述可以通过M个通道中的相应一个传送。 该编码器至少部分地基于作为中心失真和至少一个侧面失真的函数的失真度量来配置。 例如,如果M = 2,失真测量可以是形式ƒ(D0,D1,D2)的平均均方误差(AMSE)函数,其中D0是基于接收到a 第一和第二描述,D1和D2分别是仅使用第一描述和第二描述的重建产生的侧面失真。 通过扰动晶格点可以获得进一步的性能改进。 本发明的LSMDVQ技术也可以扩展到大于2的M的情况,编码器可以使用M个码本LAMBD1,LAMBD2的有序集合。 。 。 ,LAMBDM的尺寸越来越大,其中最粗的码本对应于格子。 在这种情况下,对于接收到的每个k个描述,可以存在将接收到的矢量映射到码本LAMBDk中的相应一个的单个解码功能,使得该信号的重建需要不超过M个这样的解码功能。