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公开(公告)号：US5806026A
公开(公告)日：1998-09-08
申请号：US757817
申请日：1996-11-27
申请人： Hee-Su Kim
发明人： Hee-Su Kim
CPC分类号： G10L19/0208 , G10L19/10
摘要： A degrouping method for an MPEG 1 decoder for degrouping three consecutive subband samples (X, Y and Z) compressed into one codeword .COPYRGT. by a step number (N) includes the steps determining whether the value of the step number is 3, determining whether the value of the step number is 5 if the value of the step number is not 3, determining whether the value of the step number is 9 if the value of the step number is not 5, searching corresponding values of the subband samples from a first look-up table in the sequence of Z, Y and X, if the value of the step number is 3, searching corresponding values of the subband samples from a second look-up table in the sequence of Z, Y and X, if the value of the step number is 5, and searching corresponding values of the subband samples from a third look-up table in the sequence of Z, Y and X, if the value of the step number is 9, wherein the first, second and third look-up tables have the respective values of the subband samples corresponding to the codeword value. Since the MPEG 1 degrouping method obtains subband samples using look-up tables without using a divider, the number of required cycles is considerably reduced.
摘要翻译： 用于对压缩成一个码字并复制和编号（N）的三个连续子带样本（X，Y和Z）进行分组的MPEG1解码器的去分组方法包括以下步骤：确定步骤号的值是否为3， 如果步数的值不为3，则步数的值为5，如果步数的值不为5，则确定步号的值是否为9，从第一次查找子带样本的相应值 在Z，Y和X的序列中的表，如果步数的值为3，则从Z，Y和X的序列中的第二查找表中搜索子带样本的对应值，如果值 步骤号为5，并且如果步数的值为9，则从Z，Y和X的序列中的第三查找表中搜索子带样本的相应值，其中第一，第二和第三外观 这些表具有相应的子带样本的相应值 码字值。 由于MPEG1去组合方法使用查找表获得子带样本而不使用分频器，所以需要的周期数大大减少。
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82.发明授权Dispersed impulse generator system and method for efficiently computing an excitation signal in a speech production model 失效分散脉冲发生器系统和方法，用于在语音制作模型中有效地计算激励信号
公开(公告)号：US5778337A
公开(公告)日：1998-07-07
申请号：US643522
申请日：1996-05-06
申请人： Mark A. Ireton
发明人： Mark A. Ireton
CPC分类号： G10L19/12
摘要： A vocoder for generating speech from a plurality of stored speech parameters which computes the excitation signals in the speech production model. The present invention generates a periodic excitation signal with flat frequency response and linear group delay. The present invention uses properties of the phase delay sequence being generated to calculate each of the parameters of the excitation signal in an efficient and optimized manner. Generation of the excitation signal requires computation of the expression: ##EQU1## The above expression uses the equation: ##EQU2## This equation defines the phase relationship between the signals using a linear group delay where .phi.'.sub.I (x)* is the absolute phase offset from the first phase harmonic, I is an index for the harmonic, x is time, P is the pitch period, and k" is a constant. The present invention performs the following iterations to compute the above sequence: 1) .phi.'.sub.I (x)*=.phi.'.sub.I- (x)*+A.sub.I-1 (x) 2) A.sub.I (x)=A.sub.I-1 (x)-B where A.sub.1 values are the relative phase differences between consecutive harmonics; the .phi.'.sub.I (x)* values are the absolute phase offsets from the first phase harmonic; B is a constant of 2 k"/P.sup.2, x is the time, and I is the iteration number. After the phase offset values have been computed, cosines of the plurality of phase offset values are computed and summed to produce the excitation signal. The excitation signal is then used in a speech production model to generate speech.
摘要翻译： 一种声码器，用于从多个存储的语音参数中产生语音，该语音参数计算语音产生模型中的激励信号。 本发明产生具有平坦频率响应和线性组延迟的周期性激励信号。 本发明使用生成的相位延迟序列的属性，以有效和优化的方式计算激励信号的每个参数。 激励信号的产生需要计算表达式：
（1）上述表达式使用以下等式：（2）该方程定义了使用线性组延迟的信号之间的相位关系，其中phi'I ）*是与第一相位谐波的绝对相位偏移，I是谐波的索引，x是时间，P是音调周期，k“是常数。 本发明进行以下迭代以计算上述序列：1）phi'I（x）* = phi'I（x）* + AI-1（x）2）AI（x）= AI-1 ）-B其中A1值是连续谐波之间的相对相位差; phi'I（x）*值是来自第一相位谐波的绝对相位偏移; B是2 k“/ P2的常数，x是时间，I是迭代次数。 在计算了相位偏移值之后，计算并相加多个相位偏移值的余弦以产生激励信号。 然后，激发信号用于语音产生模型中以产生语音。 -
83.发明授权
公开(公告)号：US5758021A
公开(公告)日：1998-05-26
申请号：US943307
申请日：1992-09-10
申请人： Heidi Hackbarth
发明人： Heidi Hackbarth
CPC分类号： G10L15/34 , G10L15/063 , G10L15/12 , G10L15/16
摘要： Recognition of speech with successive expansion of a reference vocabulary, can be used for automatic telephone dialing by voice input. Neural and conventional recognition methods are performed in parallel so that during training and configuration of the neural network, a conventional recognizer operating according to the dynamic programming principle has available newly added word patterns as references for immediate use in recognition. Upon completion of the training and configuration, the neural network takes over the recognition of the now expanded vocabulary.
摘要翻译： 通过连续扩展参考词汇识别语音，可用于通过语音输入进行自动电话拨号。 并行执行神经和常规识别方法，使得在神经网络的训练和配置期间，根据动态规划原理操作的常规识别器具有新添加的字模式作为在识别中立即使用的参考。 完成训练和配置后，神经网络接管现在扩展的词汇表的识别。
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公开(公告)号：US5745873A
公开(公告)日：1998-04-28
申请号：US821509
申请日：1997-03-21
申请人： Louis D. Braida , Paul Duchnowski
发明人： Louis D. Braida , Paul Duchnowski
CPC分类号： G10L15/00 , G10L2015/085
摘要： A method for recognizing speech elements (e.g., phones) in utterances includes the following steps. Based on acoustic frequency, at least two different acoustic representatives are isolated for each of the utterances. From each acoustic representative, tentative decision information on the speech element in the corresponding utterance is derived. A final decision on the speech element in the utterance is then generated, based on the tentative decision information derived from more than one of the acoustic representatives.
摘要翻译： 语音元素（例如，电话）的语音识别方法包括以下步骤。 基于声学频率，每个话语隔离出至少两个不同的声学代表。 从每个声学代表，导出相应话语中关于语音元素的暂定决定信息。 然后，基于从多个声学代表得到的暂定决定信息，产生关于话语中的语音元素的最终决定。
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公开(公告)号：US5724526A
公开(公告)日：1998-03-03
申请号：US573400
申请日：1995-12-15
申请人： Hisao Kunita
发明人： Hisao Kunita
CPC分类号： G10L15/26
摘要： An electronic interpreting machine comprising vocal input device for vocal input of a language, dictionary device incorporating an input language dictionary and an output language dictionary, language setting device for setting the input language as the first language and the output language as the second language, voice recognition device for recognizing and storing the first language, translating device for translating the first language which has been recognized into the selected second language, voice information generating device for generating voice information representing the translated second language, and voice output device for giving output of the voice information, wherein the maximum amount of language information which can be vocally input is set beforehand, while information volume computing device that computes the ratio of the amount of information which has been input to the vocal input device to the maximum amount of information which can be input in real time, and computed information notifying device for notifying the result of computation are provided.
摘要翻译： 一种电子翻译机，包括用于语言的声音输入的声音输入装置，包含输入语言字典和输出语言字典的字典装置，用于将输入语言设置为第一语言的语言设置装置和作为第二语言的输出语言 用于识别和存储第一语言的识别装置，用于将已被识别的第一语言翻译成所选择的第二语言的翻译装置，用于生成表示翻译的第二语言的语音信息的语音信息产生装置，以及用于输出 语音信息，其中可以预先设置可以被语音输入的语言信息的最大量，而信息量计算设备将已经输入到声音输入设备的信息量的比率计算为可以 实时输入，和 提供了用于通知计算结果的计算信息通知装置。
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公开(公告)号：US5704000A
公开(公告)日：1997-12-30
申请号：US337595
申请日：1994-11-10
CPC分类号： G10L25/90
摘要： A pitch estimating method includes the steps of (1) determining a set of pitch candidates to estimate a pitch of a digitized speech signal at each of a plurality of time instants, wherein series of these time instants define segments of the digitized speech signal; (2) constructing a pitch contour using a pitch candidate selected from each of the sets of pitch candidates determined in the first step; and (3) selecting a representative pitch estimate for the digitized speech signal segment from the set of pitch candidates comprising the pitch contour.
摘要翻译： 音调估计方法包括以下步骤：（1）确定一组音调候选以估计多个时刻中的每个时刻的数字化语音信号的音高，其中这些时刻的系列定义数字化语音信号的片段; （2）使用从第一步骤中确定的每个音调候选集中选择的音调候选来构造音调轮廓; 以及（3）从包括音调轮廓的音调候选集合中选择用于数字化语音信号段的代表音调估计。
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87.发明授权Speech recognition system and method which permits a speaker's utterance to be recognized using a hidden markov model with subsequent calculation reduction 失效语音识别系统和方法，允许使用隐马尔可夫模型识别说话者的话语，随后进行计算减少
公开(公告)号：US5649056A
公开(公告)日：1997-07-15
申请号：US195845
申请日：1994-02-14
申请人： Tsuneo Nitta
发明人： Tsuneo Nitta
CPC分类号： G09B19/04 , G10L15/142 , G10L15/144 , G10L25/12
摘要： A sound analyzer sound analyzes an input speech signal to obtain feature vectors. A matrix quantizer performs a matrix quantization process between the feature vectors obtained by the sound analyzer and a phonetic segment dictionary prepared in phonetic segment units to obtain a phonetic segment similarity sequence. A PS-phoneme integrating section integrates the phonetic segment similarity sequence into a phonemic feature vector. A HMM recognizer checks the phonemic feature vector using a HMM prepared in certain units, to thereby perform a recognition process.
摘要翻译： 声音分析器声音分析输入语音信号以获得特征向量。 矩阵量化器在由声音分析器获得的特征向量和以语音段单位准备的语音段字典之间执行矩阵量化处理，以获得语音段相似性序列。 PS音素整合部分将语音段相似性序列整合到音素特征向量中。 HMM识别器使用以某些单位准备的HMM来检查音素特征向量，从而执行识别过程。
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88.发明授权Methods and apparatus for noise conditioning in digital speech compression systems using linear predictive coding 失效使用线性预测编码的数字语音压缩系统中噪声调理的方法和装置
公开(公告)号：US5642464A
公开(公告)日：1997-06-24
申请号：US433116
申请日：1995-05-03
申请人： H.S. Peter Yue , Rafi Rabipour
发明人： H.S. Peter Yue , Rafi Rabipour
CPC分类号： G10L19/012 , G10L19/06
摘要： In methods and apparatus for processing a speech signals comprising a plurality of successive signal intervals, each signal interval containing no speech sounds is classified as a noise interval, and LPC coefficients are calculated for each noise interval based on the samples of that noise interval and on the samples of a plurality of preceding signal intervals. When noise intervals encoded using LPC coefficients calculated as described above are reconstructed, the subjectively annoying "swishing" or "waterfall" effects encountered in conventional LPC speech processing systems are reduced or eliminated.
摘要翻译： 在用于处理包括多个连续信号间隔的语音信号的方法和装置中，不包含语音的每个信号间隔被分类为噪声间隔，并且基于该噪声间隔的采样对每个噪声间隔计算LPC系数，并且基于 多个先前信号间隔的采样。 当使用如上所述计算的LPC系数进行编码的噪声间隔被重构时，在常规的LPC语音处理系统中遇到的主观上令人讨厌的“扫荡”或“瀑布”效应被减少或消除。
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公开(公告)号：US5577162A
公开(公告)日：1996-11-19
申请号：US320380
申请日：1994-10-11
申请人： Yasushi Yamazaki
发明人： Yasushi Yamazaki
CPC分类号： G10L15/12
摘要： A dynamic programming or DP matching system for speech recognition. Upon DP matching, cumulative distance is compared with a threshold value at every sampling time point of a speech pattern to thereby restrict the number of DP paths in succeeding matching processes. The number of DP paths remaining at each speech pattern sampling time point is monitored by a monitoring module for altering a threshold value so as to decrease the number of the DP paths. When DP path number becomes excessively large, the threshold is increased to thereby decrease the DP path number. Capacity of a DP data storing memory can be reduced while preventing the matching capability from being lowered.
摘要翻译： 用于语音识别的动态编程或DP匹配系统。 在DP匹配时，将累积距离与语音模式的每个采样时间点处的阈值进行比较，从而限制后续匹配处理中的DP路径的数量。 在每个语音模式采样时间点处剩余的DP路径的数量由用于改变阈值的监视模块监视，以便减少DP路径的数量。 当DP路径数变得过大时，阈值增大，DP路径数减少。 可以减少DP数据存储存储器的容量，同时防止匹配能力降低。
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公开(公告)号：US5396577A
公开(公告)日：1995-03-07
申请号：US994113
申请日：1992-12-22
申请人： Yoshiaki Oikawa , Kenzo Akagiri
发明人： Yoshiaki Oikawa , Kenzo Akagiri
CPC分类号： G10L13/08 , G10L13/047
摘要： In a speech synthesizing apparatus, importance degree information indicative of a degree of importance with respect to each text portion of input original text data is added to this text portion. Then, the original text data with such importance degree information is input. When a rapid reading process, or a head searching process is carried out for the original text input, speech synthesis is carried out by controlling several stages which text portion should be skipped, or at which speed, the text portions should be synthesized, in response to a speed instruction and importance degree information which are being input into the speech synthesizing apparatus.
摘要翻译： 在语音合成装置中，向该文本部分添加表示相对于输入原始文本数据的每个文本部分的重要程度的重要程度信息。 然后，输入具有这种重要度信息的原始文本数据。 当对原始文本输入进行快速读取处理或头部搜索处理时，通过控制几个阶段来执行语音合成，该阶段应当跳过文本部分，或者应该以什么速度合成文本部分 涉及正在输入到语音合成装置中的速度指令和重要度信息。
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