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    • 1. 发明授权
    • Variable dimension vector quantization
    • 可变维矢量量化
    • US5890110A
    • 1999-03-30
    • US411436
    • 1995-03-27
    • Allen GershoAmitava DasAjit Venkat Rao
    • Allen GershoAmitava DasAjit Venkat Rao
    • G10L19/00G10L19/02G10L5/06H04B1/66
    • G10L19/02
    • A variable dimension vector quantization method that uses a single "universal" codebook. The method can be given the interpretation of sampling full-dimensioned codevectors in the universal codebook and generating subcodevectors of the same dimension as input data subvector, which dimension may vary in time. A subcodevector is selected from the codebook to have minimum distortion between it and the input data subvector. The subcodevector with minimum distortion corresponds to the representative, full-dimensioned codevector in the codebook. The codebook is designed by inverse sampling of training subvectors to obtain full-dimension vectors, then iteratively clustering the training set until a stable centroid vector is obtained.
    • 使用单个“通用”码本的可变维度向量量化方法。 该方法可以给出通用码本中采样全尺寸代码矢量的解释,并生成与输入数据子向量相同维度的子代码矢量,该维度可能随时间变化。 从码本中选择一个子代码向量,使其与输入数据子向量之间具有最小的失真。 具有最小失真的子码矢量对应于码本中的代表性的,全尺寸的码矢量。 该码本是通过对训练子向量进行逆采样来设计的,以获得全维向量,然后迭代地聚类训练集,直到获得稳定的质心向量。
    • 4. 发明授权
    • Vector excitation speech or audio coder for transmission or storage
    • 用于传输或存储的矢量激励语音或音频编码器
    • US4868867A
    • 1989-09-19
    • US35518
    • 1987-04-06
    • Grant DavidsonAllen Gersho
    • Grant DavidsonAllen Gersho
    • G10L19/00G10L19/10
    • G10L19/10G10L25/06
    • A vector excitation coder compresses vectors by using an optimum codebook designed off line, using an initial arbitrary codebook and a set of speech training vectors exploiting codevector sparsity (i.e., by making zero all but a selected number of samples of lowest amplitude in each of N codebook vectors). A fast-search method selects a number N.sub.c of good excitation vectors from the codebook, where N.sub.c is much smaller thaORIGIN OF INVENTIONThe invention described herein was made in the performance of work under a NASA contract, and is subject to the provisions of Public Law 96-517 (35 USC 202) under which the inventors were granted a request to retain title.
    • 矢量激励编码器使用初始任意码本和利用码矢量稀疏性的一组语音训练矢量(即,通过使N中的每一个中的所有选定数量的最低幅度的采样数除零之外,通过使用离线设计的最佳码本来压缩向量 码本矢量)。 快速搜索方法从码本中选择Nc个良好的激励矢量,其中Nc远小于N,并且在穷举搜索中仅使用Nc向量来感知加权的输入向量zn与估计之间的最佳匹配 zn从通过长期和短期滤波器处理的码本向量导出,以及感知加权滤波器。 这些级联滤波器的零输入响应被计算,并且在感知加权之后从输入语音矢量sn中减去以产生向量rn。 使用通过计算快速内积的分子并通过用于每个码本向量cj的快速内积计算分母来执行码本搜索操作,计算方程的右侧一次一帧,然后乘法乘法 通过确定N1D2> N2D1来确定N2 / D2是否小于N1 / D1的分子和分母。 如果N2和D2不是在寄存器En和Ed中替换N1和D1。
    • 5. 发明授权
    • Frequency domain postfiltering for quality enhancement of coded speech
    • 频域后置滤波,用于编码语音的质量增强
    • US06941263B2
    • 2005-09-06
    • US09896062
    • 2001-06-29
    • Hong WangVladimir CupermanAllen GershoHosam A. Khalil
    • Hong WangVladimir CupermanAllen GershoHosam A. Khalil
    • G10L19/02G10L11/00G10L19/14G10L21/02H03M7/30
    • G10L19/26G10L21/0364
    • A method and system of performing postfiltering in the frequency domain to improve the quality of a speech signal, especially for synthesized speech resulting from codecs of low bit-rate, is provided. The method comprises LPC tilt computation and compensation methods and modules, a formant filter gain computation method and module, and an anti-aliasing method and module. The formant filter gain calculation employs an LPC representation, an all-pole modeling, a non-linear transformation and a phase computation. The LPC used for deriving the postfilter may be transmitted from an encoder or may be estimated from a synthesized or other speech signal in a decoder or receiver. The invention may be implemented in a linked decoder and encoder. A separate LPC evaluation unit that is responsible for processing and or deriving the LPC may be implemented within the invention.
    • 提供了一种在频域中执行后置滤波以提高语音信号质量的方法和系统,特别是对于由低比特率的编解码器产生的合成语音。 该方法包括LPC倾斜计算和补偿方法和模块,共振峰滤波器增益计算方法和模块,以及抗混叠方法和模块。 共振峰滤波器增益计算采用LPC表示,全极建模,非线性变换和相位计算。 用于导出后置滤波器的LPC可以从编码器发送,或者可以从解码器或接收机中的合成的或其他语音信号来估计。 本发明可以在链接的解码器和编码器中实现。 负责处理和推导LPC的单独的LPC评估单元可以在本发明中实现。
    • 10. 发明授权
    • Method for coding speech and music signals
    • 语音和音乐信号编码方法
    • US06658383B2
    • 2003-12-02
    • US09892105
    • 2001-06-26
    • Kazuhito KoishidaVladimir CupermanAmir H. MajidimehrAllen Gersho
    • Kazuhito KoishidaVladimir CupermanAmir H. MajidimehrAllen Gersho
    • G10L1902
    • G10L19/18G10L19/0212G10L19/04
    • The present invention provides a transform coding method efficient for music signals that is suitable for use in a hybrid codec, whereby a common Linear Predictive (LP) synthesis filter is employed for both speech and music signals. The LP synthesis filter switches between a speech excitation generator and a transform excitation generator, in accordance with the coding of a speech or music signal, respectively. For coding speech signals, the conventional CELP technique may be used, while a novel asymmetrical overlap-add transform technique is applied for coding music signals. In performing the common LP synthesis filtering, interpolation of the LP coefficients is conducted for signals in overlap-add operation regions. The invention enables smooth transitions when the decoder switches between speech and music decoding modes.
    • 本发明提供一种对于适用于混合编解码器的音乐信号有效的变换编码方法,由此对语音和音乐信号采用公共的线性预测(LP)合成滤波器。 LP合成滤波器分别根据语音或音乐信号的编码在语音激励发生器和变换激励发生器之间切换。 对于编码语音信号,可以使用传统的CELP技术,同时应用新的非对称重叠加法变换技术来编码音乐信号。 在执行公共LP合成滤波时,对重叠运算区域中的信号进行LP系数的插值。 当解码器在语音和音乐解码模式之间切换时,本发明实现平滑过渡。