公开(公告)号：US5307441A

公开(公告)日：1994-04-26

申请号：US442830

申请日：1989-11-29

申请人： Forrest F.-T. Tzeng

发明人： Forrest F.-T. Tzeng

IPC分类号： G10L11/02 ,  G10L19/00 ,  G10L19/08 ,  G10L19/10 ,  G10L19/12 ,  G10L19/14 ,  G10L9/02

CPC分类号： G10L19/002 ,  G10L19/083 ,  G10L19/10 ,  G10L19/12 ,  G10L19/26 ,  G10L25/78 ,  G10L25/24

摘要： A speech codec operating at low data rates uses an iterative method to jointly optimize pitch and gain parameter sets. A 26-bit spectrum filter coding scheme may be used, involving successive subtractions and quantizations. The codec may preferably use a decomposed multipulse excitation model, wherein the multipulse vectors used as the excitation signal are decomposed into position and amplitude codewords. Multipulse vectors are coded by comparing each vector to a reference multipulse vector and quantizing the resulting difference vector. An expanded multipulse excitation codebook and associated fast search method, optionally with a dynamically-weighted distortion measure, allow selection of the best excitation vector without memory or computational overload. In a dynamic bit allocation technique, the number of bits allocated to the pitch and excitation signals depend on whether the signals are "significant" or "insignificant". Silence/speech detection is based on an average signal energy over an interval and a minimum average energy over a predetermined number of intervals. Adaptive post-filter and the automatic gain control schemes are also provided. Interpolation is used for spectrum filter smoothing, and an algorithm is provided for ensuring stability of the spectrum filter. Specially designed scalar quantizers are provided for the pitch gain and excitation gain.

摘要翻译： 以低数据速率操作的语音编解码器使用迭代方法来共同优化音调和增益参数集。 可以使用26位频谱滤波器编码方案，包括连续减法和量化。 编解码器可以优选地使用分解的多脉冲激励模型，其中用作激励信号的多脉冲矢量被分解成位置和幅度码字。 通过将每个矢量与参考多脉冲矢量进行比较并量化得到的差矢量来编码多脉冲矢量。 扩展的多脉冲激励码本和相关联的快速搜索方法，可选地具有动态加权失真测量，允许选择最佳的激励矢量而无需存储器或计算过载。 在动态位分配技术中，分配给音调的位数和激励信号取决于信号是“重要”还是“不重要”。 静音/语音检测基于间隔上的平均信号能量和超过预定数量的间隔的最小平均能量。 还提供自适应后置滤波器和自动增益控制方案。 插值用于频谱滤波平滑，并提供了一种用于确保频谱滤波器稳定性的算法。 针对音调增益和激励增益提供了专门设计的标量量化器。

公开(公告)号：US5293449A

公开(公告)日：1994-03-08

申请号：US905239

申请日：1992-06-29

申请人： Forrest F. Tzeng

发明人： Forrest F. Tzeng

IPC分类号： G10L11/06 ,  G10L19/10 ,  G10L19/12 ,  G10L9/14

CPC分类号： G10L19/12 ,  G10L19/10 ,  G10L25/24 ,  G10L25/93

摘要： A linear predictive speech codec arrangement including: a spectrum synthesizer for providing reconstructed speech generation in response to excitation signals; a distortion analyzer for comparing the reconstructed speech with an original speech, and providing a distortion analysis signal in response to such comparison; and an excitation model circuit for providing excitation signals to the spectrum synthesizer, with the excitation model circuit receiving and utilizing the distortion analysis signal in an analysis-by-synthesis operation, for determining ones of excitation signals which provide an optimal reconstructed speech. The excitation model circuit can include: a voiced excitation generator and a Gaussian noise generator, both of which should optimally provide a plurality of available excitation signal models. The voiced excitation generator and Gaussian noise generator can be in the form of a codebook of a plurality of possible pulse trains and Gaussian sequences, respectively, or alternatively, the voiced excitation generator can be in the form of a first order pitch synthesizer. The optimal excitation signal and/or the pitch value and the pitch filter coefficient are determined using an analysis-by-synthesis technique.

摘要翻译： 一种线性预测语音编解码器装置，包括：频谱合成器，用于响应于激励信号提供重构的语音产生; 用于将重构语音与原始语音进行比较的失真分析器，并响应于这种比较提供失真分析信号; 以及用于向频谱合成器提供激励信号的激励模型电路，激励模型电路在分析合成操作中接收和利用失真分析信号，以确定提供最佳重构语音的激励信号。 励磁模型电路可以包括：有声激励发生器和高斯噪声发生器，它们都应该最佳地提供多个可用的激励信号模型。 有声激发发生器和高斯噪声发生器可以分别是多个可能的脉冲串和高斯序列的码本的形式，或者备选地，有声激励发生器可以是一阶音调合成器的形式。 使用分析合成技术来确定最佳激励信号和/或音调值和音调滤波器系数。

公开(公告)号：US5097507A

公开(公告)日：1992-03-17

申请号：US455047

申请日：1989-12-22

申请人： Richard L. Zinser ,  Steven R. Koch ,  Raymond L. Toy

发明人： Richard L. Zinser ,  Steven R. Koch ,  Raymond L. Toy

IPC分类号： H03M13/00 ,  G10L19/00 ,  G10L19/10 ,  H04B1/66 ,  H04B7/26 ,  H04B14/04 ,  H04L1/00 ,  H04L1/20

CPC分类号： G10L19/005 ,  H04B1/66 ,  H04L1/20 ,  H04L1/201 ,  G10L19/10

摘要： Protection of a digital multi-pulse speech coder from fading pattern bit errors common in a digital mobile radio channel is accomplished with error detection techniques which are simple to implement and require no error correcting codes. A synthetic regeneration algorithm is employed which uses only the perceptually significant bits in the transmitted frame. Separate parity checksums for line spectrum pair frequency data, pitch lag data and pulse amplitude data are added to each frame of speech coder bits in the transmitter. The bits are then transmitted through a mobile environment susceptible to fading that induces bursty error patterns in the stream. At the receiving station, the parity checksum bits and speech coder bits are used to determine if an error has occurred in a particular section of the bit stream. Detected errors are flagged and supplied to the speech decoder. The speech decoder uses the error flags to modify its output signal so as to minimize perceptual artifacts in the output speech. Separate checksums are developed for subsets of line spectrum pair (LSP) coefficients and related speech data, whereby a single subset may be error-detected and replaced, rather than an entire frame.

摘要翻译： 数字多脉冲语音编码器防止数字移动无线电信道中常见的衰落模式位错误的保护是通过易于实现的错误检测技术实现的，并且不需要纠错码。 采用合成再生算法，其仅使用传输帧中的感知有效位。 将线谱对频率数据，音调滞后数据和脉冲幅度数据的独立奇偶校验和添加到发送器中的每个语音编码器位。 然后，这些位通过易于衰落的移动环境传输，从而在流中引起突发错误模式。 在接收站，使用奇偶校验和比特和语音编码器比特来确定比特流的特定部分中是否发生错误。 检测到的错误被标记并提供给语音解码器。 语音解码器使用错误标志来修改其输出信号，以便最小化输出语音中的感知伪像。 针对线谱对（LSP）系数和相关语音数据的子集开发了单独的校验和，由此可以对单个子集进行错误检测和替换，而不是整个帧。

公开(公告)号：US5001759A

公开(公告)日：1991-03-19

申请号：US414643

申请日：1989-09-27

申请人： Akira Fukui

发明人： Akira Fukui

IPC分类号： G10L19/04 ,  G10L19/08 ,  G10L19/10

CPC分类号： G10L19/10

摘要： A multi-pulse speech coding method and apparatus capable of encoding speech at a bit rate of 16 kbps or less. The method determines the location and amplitude of a pulse by searching through all of the samples of a criterion function, modifying all of the samples of the criterion function, and them repeating the pulse search. After the predetermined number of pulses have been determined, the method modifies the amplitude of the determined pulse, modifies the criterion function at the location where the pulses are set, and repeats such pulse amplitude modification. The method is, therefore, capable of modifying a pulse amplitude by using only a minimum amount of computation. As compared to the amount of computerization required by a method of the kind which modifies pulse amplitude in a pulse search loop.

摘要翻译： 能够以16kbps以下的比特率对语音进行编码的多脉冲语音编码方法和装置。 该方法通过搜索标准函数的所有样本来确定脉冲的位置和幅度，修改标准函数的所有样本，并重复脉冲搜索。 在确定了预定数量的脉冲之后，该方法修改所确定的脉冲的幅度，修改设置脉冲的位置处的标准函数，并且重复这样的脉冲幅度修改。 因此，该方法能够通过仅使用最小量的计算来修改脉冲幅度。 与在脉冲搜索循环中修改脉冲幅度的方法所需的计算机化量相比。

公开(公告)号：US4964169A

公开(公告)日：1990-10-16

申请号：US310464

申请日：1989-02-15

申请人： Shigeru Ono

发明人： Shigeru Ono

IPC分类号： G10L19/04 ,  G10L19/10

CPC分类号： G10L19/10

摘要： A low bit rate speech coding method and implementing apparatus in which a linear predictive coding (LPC) speech synthesizer receives an excitation sequence comprised of pulses having selected amplitudes at predetermined positions within a frame to minimize the weighted mean square error between the synthetic speech produced by the LPC synthesizer and the input speech. Pulse locations and the pulse amplitudes at the respective locations are determined by a sequential processing technique in which the amplitude and location of each pulse are determined in accordance with the previously determined amplitudes and locations of the pulses preceeding the present pulse in the same frame; and specifically by determining the amplitude g.sub.k and location l.sub.k of a new pulse in a frame from selected pulses S at locations k-1 through k-S close to location l.sub.k. The number S of preceeding pulses used to determine the pulse at location l.sub.k is selected such that the distance between the Sth pulse preceeding the pulse at location l.sub.k affects the determination of the pulse at l.sub.k while pulses prior to the Sth pulse have no appreciable effect on the determination of the pulse at l.sub.k. That is, each of the S pulses within a threshold distance T.sub.th is judged to effect the detection of the pulse at l.sub.k while pulses preceeding the pulse at l.sub.k and outside of the range T.sub.th are judged to not effect the determination of the pulse at l.sub.k.

摘要翻译： 一种低比特率语音编码方法和实现装置，其中线性预测编码（LPC）语音合成器接收包括在帧内的预定位置处具有选定幅度的脉冲的激励序列，以最小化由合成语音产生的合成语音之间的加权均方误差 LPC合成器和输入语音。 脉冲位置和相应位置处的脉冲幅度由顺序处理技术确定，其中每个脉冲的幅度和位置根据先前确定的在同一帧中的当前脉冲之前的脉冲的幅度和位置来确定; 并且具体地通过从靠近位置lk的位置k-1到k-S处的选定脉冲S确定帧中的新脉冲的振幅gk和位置lk。 选择用于确定位置lk处的脉冲的前一脉冲的数量S，使得在位置lk之前的脉冲之前的第S个脉冲之间的距离影响在lk处的脉冲的确定，而在第S个脉冲之前的脉冲对于 确定脉搏在lk。 也就是说，阈值距离Tth内的S个脉冲中的每一个被判断为在lk处检测到脉冲，而在lk之前的脉冲和在范围Tth之外的脉冲被判定为不影响在lk处的脉冲的确定。

公开(公告)号：US4945567A

公开(公告)日：1990-07-31

申请号：US462981

申请日：1990-01-10

申请人： Kazunori Ozawa

发明人： Kazunori Ozawa

IPC分类号： G10L19/10

CPC分类号： G10L19/10

摘要： A method and implementing apparatus for low-bit rate speech band signal coding. An input signal in the speech band is represented by a pulse excitation sequence and a spectral parameter sequence over a frame of predetermined frame length using a selected one of a plurality of pulse determining processing modes. The selected pulse determining processing mode sequentially determines the amplitudes g.sub.i and locations m.sub.i of the pulses of the pulse excitation sequence on the basis of the amplitudes and locations of pulses in a previous frame. The selection process of determining which of the pulse determined processing modes to be used involves analyzing the input signal to produce a judgment signal d signifying the input signal as a voiced or an unvoiced signal, and selecting the pulse determining processing mode in response to the judgment signal d. The pulse excitation sequence and spectral parameter sequence are coded for transmission to a suitable receiver. The judgment signal d may also be coded and transmitted to the receiver. The receiver reproduces the input signal from the received coded signal.

摘要翻译： 一种用于低比特率语音频带信号编码的方法和实现装置。 语音频带中的输入信号使用多个脉冲确定处理模式中的所选择的一个，由预定帧长度的帧上的脉冲激励序列和频谱参数序列表示。 所选择的脉冲确定处理模式基于前一帧中的脉冲的幅度和位置顺序地确定脉冲激励序列的脉冲的振幅gi和位置mi。 确定要使用的脉冲确定处理模式中的哪一个的选择过程包括分析输入信号以产生表示输入信号作为有声或无声信号的判断信号d，并响应于判断选择脉冲确定处理模式 信号d。 脉冲激励序列和频谱参数序列被编码以传输到合适的接收机。 判断信号d也可以被编码并发送给接收机。 接收机从接收到的编码信号中再现输入信号。

公开(公告)号：US4945565A

公开(公告)日：1990-07-31

申请号：US751818

申请日：1985-07-05

申请人： Kazunori Ozawa ,  Takashi Araseki

发明人： Kazunori Ozawa ,  Takashi Araseki

IPC分类号： G10L19/10

CPC分类号： G10L19/10

摘要： In an encoder operable in response to a discrete pattern signal divisible into a succession of segments to produce an output code sequence, a pitch parameter and a spectral parameter are extracted in a parameter calculator from each segment and from a spectral interval. In an excitation pulse producing circuit, each spectral interval is divided into a plurality of subframes, namely, pitch periods with reference to the pitch parameter to divide each segment. A minor group of excitation pulses is calculated from the segment at every subframe to form a major group of the excitation pulses in the spectral interval. The excitation pulses of the major group are reduced in number with reference to adjacent ones of the minor groups in each spectral interval and are modified into a succession of modified excitation pulses. The modified excitation pulses are combined with the spectral parameter into the output code sequence. In a decoder, the modified excitation pulses and the spectral parameter are extracted from the output code sequence. The pitch parameter is recovered by the use of the extracted and mofified excitation pulses and is used to produce a reproduction of the discrete pattern signal. Alternatively, the pitch parameter may be sent from the encoder together with the spectral parameter and the modified excitation pulses as the output code sequence and extracted from the output code sequence in the decoder.

摘要翻译： 在可响应于可分成一系列段的离散模式信号进行操作以产生输出代码序列的编码器中，在参数计算器中从每个段和从频谱间隔中提取音调参数和频谱参数。 在激励脉冲产生电路中，每个频谱间隔被分为多个子帧，即，参考音调参数的音调周期，以分割每个段。 从每个子帧的段计算一小组激励脉冲，以在频谱间隔中形成主要的激励脉冲组。 主要组的激励脉冲参考每个频谱间隔中相邻的次组减少数量，并被修改成一系列修改的激励脉冲。 修改的激励脉冲与光谱参数组合成输出代码序列。 在解码器中，从输出代码序列中提取修改的激励脉冲和频谱参数。 音调参数通过使用提取和修正的激励脉冲来恢复，并用于产生离散模式信号的再现。 或者，音调参数可以与编辑器一起发送，作为输出代码序列并从解码器中的输出代码序列中提取的频谱参数和修改的激励脉冲。

公开(公告)号：US4932061A

公开(公告)日：1990-06-05

申请号：US841906

申请日：1986-03-20

申请人： Peter Kroon ,  Edmond F. A. Deprettere ,  Robert J. Sluyter

发明人： Peter Kroon ,  Edmond F. A. Deprettere ,  Robert J. Sluyter

IPC分类号： G01L9/14 ,  G10L19/10

CPC分类号： G10L19/10

摘要： A multi-pulse excitation linear-predictive speech coder operates in accordance with an analysis-by-synthesis method for determining the excitation. The coder (10) comprises an LPC-analyzer (11), a multi-phase excitation generator (13), means (12, 14) for forming an error signal representative of the difference between an original speech signal (s(n)) and a synthetic speech signal (s(n)), a filter (15) for perceptually weighting the error signal and means (16) responsive to the weighted error signal (e(n)) for generating pulse parameters controlling the excitation generator (13) so as to minimize a predetermined measure of the weighted error signal. The LPC-parameters and the pulse parameters of the excitation signal (x(n)) are encoded for efficient storage or transmission. The bit capacity required for pulse position encoding of the excitation signal (x(n)) is considerably reduced by arranging the excitation generator (16) for an excitation signal (x(n)) which in each excitation interval (L) consists of a pulse pattern having a grid of a predetermined number (q) of equidstant pulses and by arranging the control means (16) for generating pulse parameters characterizing the grid position (k) relative to the beginning of the excitation interval (L) and the variable amplitudes (b.sub.k (j), 1.ltoreq.j.ltoreq.q) of the pulses of the grid.

摘要翻译： 多脉冲激励线性预测语音编码器根据用于确定激励的合成分析方法进行操作。 编码器（10）包括LPC分析器（11），多相励磁发生器（13），用于形成表示原始语音信号（s（n））之间的差的误差信号的装置（12,14） 以及合成语音信号（s（n）），用于感知地加权误差信号的滤波器（15）和响应于加权误差信号（e（n））的装置（16）），用于产生控制激励发生器 ），以便最小化加权误差信号的预定测量。 激励信号（x（n））的LPC参数和脉冲参数被编码用于有效的存储或传输。 激励信号（x（n））的激励信号（x（n））的激励信号（x（n））的脉冲位置编码所需的位容量大大降低，激励信号（x（n））在每个激励间隔（L） 具有预定数量（q）等距脉冲的网格的脉冲图案，并且通过布置用于产生表征网格位置（k）的相对于激励间隔（L）的开始的脉冲参数的控制装置（16）和可变幅度 （bk（j），1

公开(公告)号：US4903303A

公开(公告)日：1990-02-20

申请号：US153290

申请日：1988-02-04

申请人： Tetsu Taguchi

发明人： Tetsu Taguchi

IPC分类号： G10L19/00 ,  G10L19/08 ,  G10L19/10

CPC分类号： G10L19/10

摘要： In an encoder for encoding a speech signal having a spectrum envelope into a plurality of excitation pulses, a spectrum emphasis unit emphasizes peak components of the spectrum envelope to produce an emphasized speech signal. As a result of a spectrum emphasis operation, the emphasized speech signal has an emphasized spectrum envelope which substantially comprises a plurality of line spectra. Responsive to the emphasized speech signal, a pulse producing unit produces a plurality of excitation pulses by the use of a pulse search method.

摘要翻译： 在将具有频谱包络的​​语音信号编码成多个激励脉冲的编码器中，频谱强调单元强调频谱包络的​​峰值分量以产生强调语音信号。 作为频谱强调操作的结果，强调语音信号具有基本上包括多个线谱的强调频谱包络。 响应于强调语音信号，脉冲产生单元通过使用脉冲搜索方法产生多个激励脉冲。

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。