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1.发明授权
公开(公告)号：US5864650A
公开(公告)日：1999-01-26
申请号：US762694
申请日：1996-12-12
CPC分类号： G10L19/04
摘要： A larger number, L', of delta vectors .DELTA..sub.i (i=0, 1, 2, . . . , L'-1) than the required number L are each multiplied by a matrix of a linear predictive synthesis filter (3), their power (A.DELTA..sub.i).sup.T (A.DELTA..sub.i) is evaluated (42), and the delta vectors are reordered in decreasing order of power (43); then, L delta vectors are selected in decreasing order of power, the largest power first, to construct a tree-structure data code book (41), using which A-b-S vector quantization is performed (48). This provides increased freedom for the space formed by the delta vectors and improves quantization characteristics. Further, variable rate encoding is achieved by taking advantage of the structure of the tree-structure data code book.
摘要翻译： 每个乘以线性预测合成滤波器（3）的矩阵，将大于所需数量L的Δ向量DELTA i（i = 0,1,2，...，L'-1）的较大数量L' ，评估其功率（A DELTA i）T（A DELTA i）（42），并且以递减的功率顺序对增量向量重新排序（43）; 然后，以降序的功率，最大功率先选择L个增量向量，构建一个树结构数据码本（41），利用其中的A-b-S矢量量化进行（48）。 这提供了由Δ向量形成的空间的增加的自由度并且提高了量化特性。 此外，通过利用树结构数据码本的结构来实现可变速率编码。
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公开(公告)号：US5199076A
公开(公告)日：1993-03-30
申请号：US761048
申请日：1991-09-18
摘要： A CELP type speech coding system is provided with an arithmetic processing unit which transforms a perceptual weighted input speech signal vector AX to a vector .sup.t AAX, a sparse adaptive codebook which stores a plurality of pitch prediction residual vectors P sparsed by a sparse unit, and a multiplying unit which multiplies the successively read out vectors P and the output .sup.t AAX from the arithmetic processing unit. In addition, the CELP type speech coding system includes a filter operation unit which performs a filter operation on the vectors P, and an evaluation unit which finds the optimum vector P based on the output from the filter operation unit, so as to enable reduction of the amount of arithmetic operations.
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3.发明授权Speech coding system having codebook storing differential vectors between each two adjoining code vectors 失效语音编码系统，其具有在两个相邻码矢量之间存储差分矢量的码本
公开(公告)号：US5323486A
公开(公告)日：1994-06-21
申请号：US856221
申请日：1992-05-14
申请人： Tomohiko Taniguchi , Mark Johnson , Yasuji Ohta , Hideaki Kurihara , Yoshinori Tanaka , Yoshihiro Sakai
发明人： Tomohiko Taniguchi , Mark Johnson , Yasuji Ohta , Hideaki Kurihara , Yoshinori Tanaka , Yoshihiro Sakai
摘要： A speech coding system is provided where input speech is coded by finding via an evaluation computation a code vector giving a minimum error between reproduced signals obtained by linear prediction analysis filter processing, simulating speech path characteristics, on code vectors successively read out from a noise codebook storing a plurality of noise trains as code vectors and an input speech signal and by using a code specifying the code vector. In the speech coding system, the noise codebook includes a delta vector codebook which stores an initial vector and a plurality of delta vectors having difference vectors between adjoining code vectors. In addition, provision is made in the computing unit for the evaluation computation of a cyclic adding unit for cumulatively adding the delta vectors to virtually reproduce the code vectors.
摘要翻译： PCT No.PCT / JP91 / 01235 Sec。 371日期：1992年5月14日 102（e）日期1992年5月14日PCT 1991年9月17日PCT PCT。 第WO92 / 05541号公报 日期：1992年4月2日。提供一种语音编码系统，其中输入语音通过经由评估计算找出通过线性预测分析滤波处理获得的再现信号之间的最小误差的代码矢量，模拟语音路径特性，代码矢量 从存储多个噪声序列的噪声码本连续地读出作为码矢量和输入语音信号，并通过使用指定码矢量的码来读出。 在语音编码系统中，噪声码本包括增量向量码本，其存储初始向量和具有相邻码矢量之间的差矢量的多个Δ向量。 此外，在计算单元中进行用于循环加法单元的评估计算的计算，用于累加Δ向量以虚拟地再现代码矢量。
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4.发明授权Speech signal coding system wherein non-periodic component feedback to periodic excitation signal source is adaptively reduced 失效语音信号编码系统，其中对周期性激励信号源的非周期性分量反馈被自适应地减少
公开(公告)号：US5828811A
公开(公告)日：1998-10-27
申请号：US188124
申请日：1994-01-28
摘要： Speech coding uses a periodic excitation signal source, a non-periodic signal excitation source, a synthesizing unit, a filter, and an error evaluating unit. The synthesizing unit synthesizes a periodic excitation signal and a non-periodic excitation signal output from the above-mentioned sources to generate an excitation sound source signal. The filter regenerates an input speech signal from the excitation sound signal. The error evaluating unit controls the periodic excitation signal source and the non-periodic excitation signal source so that the above-mentioned units output periodic and non-periodic excitation signals which minimize a difference of the regenerated signal from an input speech signal. When the error is minimized, the above-mentioned sound source is supplied to a feedback excitation signal modifying unit. The feedback excitation modifying unit reduces a non-periodic component in the above-mentioned supplied signal so that the non-periodic component is reduced more when the relative amount of the periodic component is greater. The modified signal is fed back to the periodic excitation signal source unit to modify the content thereof.
摘要翻译： 语音编码使用周期性激励信号源，非周期信号激励源，合成单元，滤波器和误差评估单元。 合成单元合成从上述源输出的周期性激励信号和非周期性激励信号，以产生激发声源信号。 滤波器从激励声音信号再生输入语音信号。 误差评估单元控制周期性激励信号源和非周期激励信号源，使得上述单元输出使再生信号与输入语音信号的差异最小化的周期性和非周期性激励信号。 当误差最小化时，将上述声源提供给反馈激励信号修改单元。 反馈激励修正单元在上述提供的信号中减少非周期性分量，使得当周期性分量的相对量较大时，非周期分量更多地减小。 将修改的信号反馈到周期性激励信号源单元以修改其内容。
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公开(公告)号：US5086471A
公开(公告)日：1992-02-04
申请号：US545609
申请日：1990-06-29
CPC分类号： G10L19/083 , G06T9/008 , H03M7/3082
摘要： A gain-shape vector quantization apparatus for compressing the data of voice signal. A code book portion is constituted by a plurality of shape vectors which produce a plurality of selected shape vectors. A plurality of variable gain circuits impart gains to each shape vector produced from the code book portion. A plurality of synthesis filters regenerate signals from the outputs of the variable gain circuits. An adder adds the signals regenerated by the synthesis filters. An evaluation unit produces an index to select a plurality of shape vectors in the code book portion in order to minimize an error between the output of the adder and an input speech signal and further produces gain adjusting signal for the variable gain circuits.
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公开(公告)号：US5115469A
公开(公告)日：1992-05-19
申请号：US460099
申请日：1990-02-08
申请人： Tomohiko Taniguchi , Kohei Iseda , Koji Okazaki , Fumio Amano , Shigeyuki Unagami , Yoshinori Tanaka , Yasuji Ohta
发明人： Tomohiko Taniguchi , Kohei Iseda , Koji Okazaki , Fumio Amano , Shigeyuki Unagami , Yoshinori Tanaka , Yasuji Ohta
摘要： Several encoders perform a local decoding of a speech signal and extract excitation information and vocal tract information from a speech signal for an encoding operation. The transmission rate ratio between the excitation information and the vocal tract information are different for each encoder. An evaluation/selection unit evaluates the quality of decoded signals subjected to a local decoding in each of the encoders, determines the most suitable encoders from among the several encoders based on the result of the evaluation, and selects the most suitable encoder, thereby outputting the selection result as selection information. The decoder decodes a speech signal based on selection information, vocal tract information and excitation information. The evaluation/selection unit selects the output from the encoder in which the quality of a locally decoded signal is the most preferable. When vocal tract information changes little, the vocal tract information is not output, thereby allowing for increased quality of information. As much of the surplus of unused vocal tract information as possible is assigned to a residual signal. Thus, the quality of a decoded speech signal is improved.
摘要翻译： PCT No.PCT / JP89 / 00580 Sec。 371日期1990年2月8日 102（e）1990年2月8日PCT PCT 1989年6月7日PCT公布。 出版物WO89 / 12292 日期为1989年12月14日。几个编码器执行语音信号的本地解码，并从用于编码操作的语音信号中提取激励信息和声道信息。 激励信息和声道信息之间的传输速率比对于每个编码器是不同的。 评估/选择单元评估在每个编码器中经过本地解码的解码信号的质量，基于评估结果从多个编码器中确定最合适的编码器，并且选择最合适的编码器，从而输出 选择结果作为选择信息。 解码器基于选择信息，声道信息和激励信息来解码语音信号。 评估/选择单元选择编码器的输出，其中本地解码信号的质量是最优选的。 当声道信息变化较小时，不输出声道信息，从而允许提高信息质量。 尽可能多的未使用的声道信息的剩余被分配给残余信号。 因此，提高了解码语音信号的质量。
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公开(公告)号：US5261027A
公开(公告)日：1993-11-09
申请号：US997667
申请日：1992-12-28
申请人： Tomohiko Taniguchi , Yoshinori Tanaka , Yasuji Ohta , Fumio Amano , Shigeyuki Unagami , Akira Sasama
发明人： Tomohiko Taniguchi , Yoshinori Tanaka , Yasuji Ohta , Fumio Amano , Shigeyuki Unagami , Akira Sasama
摘要： A code excited linear prediction (CELP) type speech signal coding system is provided, a code vector obtained by applying linear prediction to a vector of a residual speech signal of white noise is stored in a code book. A pitch prediction vector obtained by applying linear prediction to a residual signal of a preceding frame is given a delay corresponding to a pitch frequency and added to the code vector. Use is made of an impulse vector obtained by applying linear prediction to a residual signal vector of impulses having a predetermined relationship with the vectors of the white noise code book. Variable gains are given to at least the above code vector and impulse vector, a reproduced signal is produced, and this reproduced signal is used for identification of the input speech signal. Thus, a pulse series corresponding to the sound source of voiced speech sounds is created.
摘要翻译： 提供码激励线性预测（CELP）型语音信号编码系统，通过对白噪声的残留语音信号的向量应用线性预测而获得的码矢量被存储在码本中。 通过对前一帧的残差信号应用线性预测而获得的音调预测矢量被给予对应于音调频率的延迟并被加到码矢量。 使用通过对与白噪声码本的矢量具有预定关系的脉冲的残差信号矢量应用线性预测而获得的脉冲矢量。 至少对上述代码矢量和脉冲矢量赋予可变增益，产生再生信号，该再现信号用于识别输入语音信号。 因此，产生对应于有声语音的声源的脉冲序列。
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公开(公告)号：US5151968A
公开(公告)日：1992-09-29
申请号：US562604
申请日：1990-08-03
CPC分类号： G10L19/038 , G06T9/008 , H03M7/3082
摘要： An apparatus for compressing and decompressing speech signals. The encoder or decoder comprises a plurality of codebooks each controlling a plurality of indexed code vectors for a different frequency band. Each of the codebooks is provided with a synthesis filter for reproducing a signal wave shape based on a code vector provided by the corresponding codebook. The encoder or decoder further comprises an adder for computing a sum of signal wave shapes reproduced by the synthesis filters. This arrangement can reduce the memory size of each codebook used for an encoding or a decoding process and an amount of operations of the encoding or decoding process.
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公开(公告)号：US5263119A
公开(公告)日：1993-11-16
申请号：US795668
申请日：1991-11-21
CPC分类号： G10L19/083 , G06T9/008 , H03M7/3082
摘要： A gain-shape vector quantization apparatus is provided for encoding and decoding, to transmit and receive compressed speech signals. A selected plurality of vectors are read from a code book based upon an index signal. The vectors are added in an adder and synthesis filtered by a synthesis filter, in either order, to produce an output. This output is subtracted from an input speech signal to produce an error signal. An evaluation unit produces an index to select the plurality of vectors read from the code book memory based on the error signal in order to minimize this error signal. The evaluation unit produces gain adjusting signals which can be used to adjust gains of the vectors read from the code book. In an encoder, signals indicative of the gain adjusting signal and the index signal are transmitted by a transmitter of the encoder to send a quantized speech signal to a receiver of a decoder. In the decoder, after the signals indicative of the gain adjusting signal and the index are received by the receiver of the decoder, an index and gain adjusting signal is derived for use to control reading of vectors from a code book and gains thereon to reproduce the speech signal.
摘要翻译： 增益形矢量量化装置被提供用于编码和解码，以发送和接收压缩的语音信号。 基于索引信号，从码本读取所选择的多个向量。 向量以加法器的形式加到合成滤波器中，并以合成滤波器的顺序进行合成，以产生输出。 从输入语音信号中减去该输出以产生误差信号。 评估单元基于误差信号产生用于选择从码本存储器读取的多个向量的索引，以使该误差信号最小化。 评估单元产生可用于调整从码本读取的向量的增益的增益调整信号。 在编码器中，指示增益调整信号和索引信号的信号由编码器的发射器发送，以将量化的语音信号发送到解码器的接收机。 在解码器中，在由解码器的接收器接收到指示增益调整信号和索引的信号之后，导出索引和增益调整信号以用于控制来自码本的向量的读取并在其上进行增益以再现 语音信号。
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公开(公告)号：US5231692A
公开(公告)日：1993-07-27
申请号：US593756
申请日：1990-10-05
申请人： Yoshinori Tanaka , Tomohiko Taniguchi , Akira Sasama , Yasuji Ohta , Fumio Amano , Shigeyuki Unagami
发明人： Yoshinori Tanaka , Tomohiko Taniguchi , Akira Sasama , Yasuji Ohta , Fumio Amano , Shigeyuki Unagami
CPC分类号： G10L25/90
摘要： Pitch periods for a long term predictor included in a speech codec are searched in two searching stages. In the first searching stage, probable pitch periods are searched skipping a constant number of pitch periods, and in the second searching stage, pitch periods including the pitch period determined in the first searching stage and pitch periods neighboring the pitch period on both sides are searched.
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