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1.发明授权
公开(公告)号：US5274741A
公开(公告)日：1993-12-28
申请号：US515430
申请日：1990-04-27
CPC分类号： G10L19/038 , G10L19/06 , H03M7/3082
摘要： A speech coding apparatus includes multipliers and prediction filters which successively process a plurality of signal vectors obtained from an index 2.sup.M and dimension N code book to obtain a reproduced speech signal. Error detectors are provided which find the error between the input speech signal and reproduced speech signal. Evaluators are also provided which calculate the optimum signal vectors giving the smallest errors. The multipliers are connected to a reduced code book, which is constituted of n number of code book blocks of index 2.sup.M/n and dimension N/n (where n is an integer of two or more). There are n number of multipliers, n number of prediction filters, n number of error detectors, and n number of evaluators corresponding to the code book blocks.
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公开(公告)号：US5138662A
公开(公告)日：1992-08-11
申请号：US508553
申请日：1990-04-13
CPC分类号： G10L19/12
摘要： A speech coding apparatus which selects an optimum code from a code book, the optimum code giving the minimum magnitude of error signal between the input signal and the reproduced signal obtained by a filter calculation using a linear prediction parameter from a linear predictive analysis unit with respect to the codes of the code book, wherein the code book is formed by thinning to 1/M (M being an integer of two or more) the plurality of sampling values constituting the codes. To compensate for the deterioration of the quality of the reproduced signal caused by thinning the sampling values in this way, an additional linear predictive analysis unit is further introduced and use made of an amended linear prediction parameter instead of the linear prediction parameter from the originally provided linear predictive analysis unit.
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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：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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公开(公告)号：US5224167A
公开(公告)日：1993-06-29
申请号：US580669
申请日：1990-09-11
申请人： Tomohiko Taniguchi , Yoshinori Tanaka , Akira Sasama , Yasuji Ohta , Fumio Amano , Shigeyuki Unagami
发明人： Tomohiko Taniguchi , Yoshinori Tanaka , Akira Sasama , Yasuji Ohta , Fumio Amano , Shigeyuki Unagami
摘要： A speech coding apparatus coupled to a transmission channel includes m (m is an integer greater than 1) coders, m decoders and m or (m-1) error correcting coders. The apparatus also includes an evaluation unit which evaluates a quality of each of reproduced speech signals from the input speech signal and the reproduced speech signals and which outputs an evaluated quality of each of the reproduced speech signals. The quality of each of the reproduced speech signals is evaluated in a state having no transmission error. A decision unit identifies one of the m coders which provides the reproduced speech signal having a smallest distortion on the basis of the evaluated quality of each of the reproduced speech signals, a current error rate of the transmission channel and error correcting abilities of the error correcting coders, and generates a coder identification number representative of a selected one of the m coders. An output part outputs a multiplexed transmission signal including the coded speech signal generated by the one of the m coders identified by the decision unit and the error correcting code generated by a corresponding one of the m error correcting coders.
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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：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：US4890325A
公开(公告)日：1989-12-26
申请号：US156882
申请日：1988-02-18
摘要： Speech is data-compressed and coded for transmission using TDHC (Time Domain Harmonic Compression) for the voiced signal, and decimated sampling for the unvoiced signal. Features include voiced/unvoiced detection, pitch period detection, border detection, coding by ADPCM, and optimum quantization coding. Reception involves decoding and data-reconstruction.
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10.发明授权Coding transmission equipment for carrying out coding with adaptive quantization 失效编码用自适应量化进行编码的传输设备
公开(公告)号：US4831636A
公开(公告)日：1989-05-16
申请号：US134818
申请日：1987-12-18
申请人： Tomohiko Taniguchi , Kohei Iseda , Yoshihiro Tomita , Fumio Amano , Shigeyuki Unagami , Shoji Tominaga
发明人： Tomohiko Taniguchi , Kohei Iseda , Yoshihiro Tomita , Fumio Amano , Shigeyuki Unagami , Shoji Tominaga
CPC分类号： H04B14/046 , G06T9/005 , H04B14/068
摘要： Coding transmission equipment wherein an plurality of adaptive coding units having different processing characteristics are provided, and the adaptive coding unit having the optimum processing characteristic for a current input signal is selected from among the plurality of adaptive coding units, and an output signal from the optimum adaptive coding unit and the unit number thereof are transmitted to the receiver side. At the receiver side, the processing characteristic of a decoder is changed to become the optimum processing characteristic indicated by the received unit number, whereby a high quality recovered signal is realized.
摘要翻译： 编码传输设备，其中提供具有不同处理特性的多个自适应编码单元，并且从多个自适应编码单元中选择具有用于当前输入信号的最佳处理特性的自适应编码单元，以及来自最佳 自适应编码单元及其单元号被发送到接收机侧。 在接收机侧，将解码器的处理特性改变为由接收到的单元号表示的最佳处理特性，从而实现高质量的恢复信号。
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