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1. 发明授权

US4457004A Multidimensional channel coding 失效
标题翻译：多维通道编码
公开(公告)号：US4457004A
公开(公告)日：1984-06-26
申请号：US346627
申请日：1982-02-08
申请人： Allen Gersho , Victor B. Lawrence
发明人： Allen Gersho , Victor B. Lawrence
IPC分类号： H04L1/00 , H04L25/48 , H04L25/49 , H04L27/34 , H04L25/04 , H04L25/20
CPC分类号： H04L1/0057 , H04L25/49 , H04L27/3427
摘要： An n-dimensional channel code is used in a data transmission system. The alphabet of codewords (data symbols) comprises a subset of points of a selected coset of a selected lattice. The alphabet includes at least one point of the coset whose norm (signal energy) is greater than at least one other point of the coset which is excluded from the alphabet.
摘要翻译：在数据传输系统中使用n维信道码。码字的字母表（数据符号）包括所选择的格子的选定陪集的点的子集。字母表包括其范围（信号能量）大于从字母表排除的陪集的至少一个其他点的陪集中的至少一个点。

2. 发明授权

US4412341A Interference cancellation method and apparatus 失效
标题翻译：干扰消除方法和装置
公开(公告)号：US4412341A
公开(公告)日：1983-10-25
申请号：US322517
申请日：1981-11-18
申请人： Allen Gersho , Edmond Y. Ho , Richard D. Gitlin , Victor B. Lawrence , Tong L. Lim
发明人： Allen Gersho , Edmond Y. Ho , Richard D. Gitlin , Victor B. Lawrence , Tong L. Lim
IPC分类号： H04L27/00 , H04B3/20 , H04B15/00 , H04L27/01 , H04B3/14
CPC分类号： H04L27/01 , H04B15/00
摘要： Weighted sums of samples of an interference-corrupted data signal are generated to form phase-compensated, symbol-spaced samples. The interference-corrupted signal is at the same time subjected to conventional adaptive equalization and is otherwise processed so as to form tentative decisions as to the transmitted data. Cancellation signals representing the interference components of the phase-compensated samples are generated by forming respective weighted sums of the tentative decisions and each cancellation signal is combined with a respective phase-compensated sample to provide cancelled samples from which final data decisions are made. The weighting coefficients used to form the phase-compensated samples and the cancellation signals are adaptively updated in response to error signals derived from the difference between each cancelled sample and the corresponding final decision.
摘要翻译：产生干扰损坏的数据信号的加权加权和，以形成相位补偿的符号间隔样本。干扰损坏的信号同时经历常规的自适应均衡，并且被处理，以便形成关于发送的数据的暂定决定。表示相位补偿样本的干扰分量的消除信号通过形成临时判定的各自的加权和来产生，并且将每个消除信号与相应的相位补偿样本组合，以提供取消的取样，从中作出最终的数据决定。用于形成相位补偿采样的加权系数和消除信号根据从每个取消的采样与相应的最终决策之间的差异导出的误差信号自适应地更新。

3. 发明授权

US07584095B2 REW parametric vector quantization and dual-predictive SEW vector quantization for waveform interpolative coding 有权
公开(公告)号：US07584095B2
公开(公告)日：2009-09-01
申请号：US11234631
申请日：2005-09-23
申请人： Oded Gottesman , Allen Gersho
发明人： Oded Gottesman , Allen Gersho
IPC分类号： G10L19/14
CPC分类号： G10L19/097
摘要： An enhanced analysis-by-synthesis waveform interpolative speech coder able to operate at 2.8 kbps. Novel features include dual-predictive analysis-by-synthesis quantization of the slowly-evolving waveform, efficient parametrization of the rapidly-evolving waveform magnitude, and analysis-by-synthesis vector quantization of the rapidly evolving waveform parameter. Subjective quality tests indicate that it exceeds G.723.1 at 5.3 kbps, and of G.723.1 at 6.3 kbps.

4. 发明授权

US07286982B2 LPC-harmonic vocoder with superframe structure 有权
标题翻译：具有超帧结构的LPC谐波声码器
公开(公告)号：US07286982B2
公开(公告)日：2007-10-23
申请号：US10894854
申请日：2004-07-20
申请人： Allen Gersho , Vladimir Cuperman , Tian Wang , Kazuhito Koishida
发明人： Allen Gersho , Vladimir Cuperman , Tian Wang , Kazuhito Koishida
IPC分类号： G10L19/12
CPC分类号： G10L19/173 , G10L19/087
摘要： An enhanced low-bit rate parametric voice coder that groups a number of frames from an underlying frame-based vocoder, such as MELP, into a superframe structure. Parameters are extracted from the group of underlying frames and quantized into the superframe which allows the bit rate of the underlying coding to be reduced without increasing the distortion. The speech data coded in the superframe structure can then be directly synthesized to speech or may be transcoded to a format so that an underlying frame-based vocoder performs the synthesis. The superframe structure includes additional error detection and correction data to reduce the distortion caused by the communication of bit errors.
摘要翻译：一种增强的低比特率参数语音编码器，其将来自基础帧的声码器（例如MELP）的多个帧分组成超帧结构。从底层帧中提取参数并量化到超帧中，这允许在不增加失真的情况下减少底层编码的比特率。然后，可以将在超帧结构中编码的语音数据直接合成为语音，或者将其转码为格式，使得基础的基于帧的声码器进行合成。超帧结构包括附加的错误检测和校正数据，以减少由位错误的通信引起的失真。

5. 发明授权

US4868867A Vector excitation speech or audio coder for transmission or storage 失效
标题翻译：用于传输或存储的矢量激励语音或音频编码器
公开(公告)号：US4868867A
公开(公告)日：1989-09-19
申请号：US35518
申请日：1987-04-06
申请人： Grant Davidson , Allen Gersho
发明人： Grant Davidson , Allen Gersho
IPC分类号： G10L19/00 , G10L19/10
CPC分类号： G10L19/10 , G10L25/06
摘要： 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.
摘要翻译：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。

6. 发明授权

US06941263B2 Frequency domain postfiltering for quality enhancement of coded speech 失效
标题翻译：频域后置滤波，用于编码语音的质量增强
公开(公告)号：US06941263B2
公开(公告)日：2005-09-06
申请号：US09896062
申请日：2001-06-29
申请人： Hong Wang , Vladimir Cuperman , Allen Gersho , Hosam A. Khalil
发明人： Hong Wang , Vladimir Cuperman , Allen Gersho , Hosam A. Khalil
IPC分类号： G10L19/02 , G10L11/00 , G10L19/14 , G10L21/02 , H03M7/30
CPC分类号： G10L19/26 , G10L21/0364
摘要： 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.
摘要翻译：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。

7. 发明授权

US06475245B2 Method and apparatus for hybrid coding of speech at 4KBPS having phase alignment between mode-switched frames 有权
标题翻译：用于具有在模式切换帧之间的相位对准的4KBPS的语音的混合编码的方法和装置
公开(公告)号：US06475245B2
公开(公告)日：2002-11-05
申请号：US09777424
申请日：2001-02-05
申请人： Allen Gersho , Eyal Shlomot , Vladimir Cuperman , Chunyan Li
发明人： Allen Gersho , Eyal Shlomot , Vladimir Cuperman , Chunyan Li
IPC分类号： G10L1106
CPC分类号： G10L19/18 , G10L19/02 , G10L19/09 , G10L19/10 , G10L2025/935
摘要： A method and apparatus for encoding speech for communication to a decoder for reproduction of the speech where the speech signal is classified into steady state voiced (harmonic), stationary unvoiced, and “transitory” or “transition” speech, and a particular type of coding scheme is used for each class. Harmonic coding is used for steady state voiced speech, “noise-like” coding is used for stationary unvoiced speech, and a special coding mode is used for transition speech, designed to capture the location, the structure, and the strength of the local time events that characterize the transition portions of the speech. The compression schemes can be applied to the speech signal or to the LP residual signal.
摘要翻译：用于将语音编码的语音的方法和装置用于再现语音的语音的解码器，其中语音信号被分类为稳态浊音（谐波），固定无声和“暂时”或“转换”语音，以及特定类型的编码方案用于每个类。谐波编码用于稳态浊音，“静噪”编码用于固定无声语音，特殊编码方式用于转换语音，用于捕获本地时间的位置，结构和强度表征演讲过渡部分的事件。压缩方案可以应用于语音信号或LP残差信号。

8. 发明授权

US4012628A Filter with a reduced number of shift register taps 失效
标题翻译：用减少数量的移位寄存器抽头进行滤波
公开(公告)号：US4012628A
公开(公告)日：1977-03-15
申请号：US605073
申请日：1975-08-15
申请人： Allen Gersho
发明人： Allen Gersho
IPC分类号： H03H17/06 , G06F7/38 , H04B3/04
CPC分类号： H03H17/06
摘要： Disclosed is a discrete-time filter configuration employing a shift register having output taps at only preselected stages for developing an output signal that is responsive to the signal of each stage of a shift register. The desired output signal is computed in a filter processing network by multiplying each of the output taps signals by a plurality of filter coefficients, by appropriately delaying the multiplied signals and by summing the delayed signals to produce the desired output signal.
摘要翻译：公开了采用移位寄存器的离散时间滤波器配置，该移位寄存器仅在预选阶段具有输出抽头，用于开发响应于移位寄存器的每个级的信号的输出信号。在滤波处理网络中通过将每个输出抽头信号乘以多个滤波器系数，通过适当地延迟相乘的信号并且对延迟的信号求和以产生期望的输出信号来计算期望的输出信号。

9. 发明申请

US20060069554A1 REW parametric vector quantization and dual-predictive SEW vector quantization for waveform interpolative coding 有权
公开(公告)号：US20060069554A1
公开(公告)日：2006-03-30
申请号：US11234631
申请日：2005-09-23
申请人： Oded Gottesman , Allen Gersho
发明人： Oded Gottesman , Allen Gersho
IPC分类号： G10L19/12
CPC分类号： G10L19/097
摘要： An enhanced analysis-by-synthesis waveform interpolative speech coder able to operate at 2.8 kbps. Novel features include dual-predictive analysis-by-synthesis quantization of the slowly-evolving waveform, efficient parametrization of the rapidly-evolving waveform magnitude, and analysis-by-synthesis vector quantization of the rapidly evolving waveform parameter. Subjective quality tests indicate that it exceeds G.723.1 at 5.3 kbps, and of G.723.1 at 6.3 kbps.

10. 发明授权

US06658383B2 Method for coding speech and music signals 有权
标题翻译：语音和音乐信号编码方法
公开(公告)号：US06658383B2
公开(公告)日：2003-12-02
申请号：US09892105
申请日：2001-06-26
申请人： Kazuhito Koishida , Vladimir Cuperman , Amir H. Majidimehr , Allen Gersho
发明人： Kazuhito Koishida , Vladimir Cuperman , Amir H. Majidimehr , Allen Gersho
IPC分类号： G10L1902
CPC分类号： G10L19/18 , G10L19/0212 , G10L19/04
摘要： 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.
摘要翻译：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。

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