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公开(公告)号：US08033569B2
公开(公告)日：2011-10-11
申请号：US12480178
申请日：2009-06-08
申请人： Mitsutoshi Yamanishi , Fumio Amano
发明人： Mitsutoshi Yamanishi , Fumio Amano
IPC分类号： B60R21/213 , B60R21/232 , B60R21/2338
CPC分类号： B60R21/2338 , B60R21/213 , B60R21/232 , B60R21/237 , B60R22/195 , B60R2021/23386
摘要： A curtain member has a bending portion which is formed by a rear portion of the curtain member in a folded stored state being bent forward, i.e., toward an opposite side to the vertically-extending vehicle-body pillar. A second tether member is made of a long cloth member. Its rear end is fixed to a specified lower-side position of a fifth pillar which is located below the middle portion of this pillar in the vertical direction via a fixing pin. Its front end is fixed to a body portion of a curtain member via a seam portion. Accordingly, there can be provided the interior structure of a vehicle equipped with a curtain airbag which can properly form the tension line at the curtain member.
摘要翻译： 帘式构件具有弯曲部，该弯曲部由折叠存储状态的帘幕构件的后部形成，向前弯曲，即朝向垂直延伸的车身柱的相对侧。 第二系绳构件由长布构件制成。 其后端通过固定销固定在位于该柱的中间部分下方的第五柱的指定的下侧位置。 其前端经由接缝部固定于帘部件的主体部。 因此，可以提供一种车辆的内部结构，该车辆配备有能够在帘部件上适当地形成张力线的帘式气囊。
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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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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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6.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：US5136577A
公开(公告)日：1992-08-04
申请号：US658180
申请日：1991-02-20
申请人： Fumio Amano , Hector M. P. Meana
发明人： Fumio Amano , Hector M. P. Meana
摘要： A sub-band acoustic echo canceller includes a first division and decimation process part for dividing a reception signal from a line into first band signals of N channels and for decimating each of the first band signals to output decimated first band signals in a form of complex signals, where N is an integer greater than or equal to two, a second division and decimation process part for dividing a transmission signal into second band signals of N channels and for decimating each of the second band signals to output decimated second band signals in a form of complex signals, an echo canceller group made up of a group of echo cancellers for generating a pseudo echo in each band based on a corresponding one of the decimated first band signals received from the first division and decimation process part by referring to a corresponding one of the decimated second band signals received from the second division and decimation process part and for outputting a residual signal in each band by subtracting the pseudo echo of one band from the decimated second band signal of the same band, and an interpolation and synthesis process part for subjecting the residual signals received from the echo canceller group to interpolation and synthesis to output a synthesized residual signal which is transmitted to a line as the transmission signal, where the echo canceller group carries out an echo cancelling operation for each band in a complex signal region.
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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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9.发明授权Echo canceller with short processing delay and decreased multiplication number 失效回波消除器具有短处理延迟和减少的倍数
公开(公告)号：US4951269A
公开(公告)日：1990-08-21
申请号：US216907
申请日：1988-06-24
CPC分类号： H04B3/23 , H03H21/00 , H03H21/0027 , Y10S367/901 , Y10S367/903
摘要： An echo canceller in a system, having a long impulse response such as in an acoustic system, which employs a fast Fourier transform to transform input data represented in the time domain into signals represented in the frequency domain to reduce calculations. To solve the problem of a long delay, the impulse response length is divided into a plurality of blocks. Each block then has a decreased number of samples within each block. Thus, a fast Fourier transform and finite impulse response type digital filtering are effected, so that the processing delay is decreased while the amount of calculations is kept small.
摘要翻译： PCT No.PCT / JP87 / 00833 Sec。 371日期1988年6月24日第 102（e）日期1988年6月24日PCT提交1987年10月29日PCT公布。 第WO88 / 03341号公报 日期：1988年5月5日。在具有诸如在声学系统中的长脉冲响应的系统中的回波消除器，其采用快速傅里叶变换来将在时域中表示的输入数据变换成在频域中表示的信号以减少计算 。 为了解决长延迟的问题，脉冲响应长度被分成多个块。 每个块在每个块内具有减少的采样数。 因此，实现快速傅里叶变换和有限脉冲响应型数字滤波，使得处理延迟减小，同时计算量保持较小。
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公开(公告)号：US06871175B2
公开(公告)日：2005-03-22
申请号：US09816032
申请日：2001-03-22
申请人： Fumio Amano
发明人： Fumio Amano
CPC分类号： G10L19/00
摘要： A voice encoding method includes the steps of encoding a first frame that contains a plurality of voice data into encoded parameters, locally decoding the encoded parameters of the first frame into a second frame, performing a plurality of interpolation recovery processes that generate respective frames approximating to the first frame by using a frame or frames other than the first frame, comparing the second frame with the frames approximating to the first frame generated by the plurality of interpolation recovery processes, calculating a signal to noise ratio of each of the frames approximating to the first frame by treating the second frame as the signal, determining an index number that indicates an interpolation recovery process which provides a highest signal to noise ratio, and multiplexing and transmitting the index number with the encoded parameters.
摘要翻译： 语音编码方法包括以下步骤：将包含多个语音数据的第一帧编码为编码参数，将第一帧的编码参数本地解码为第二帧，执行多个内插恢复处理，该多个内插恢复处理生成近似于 通过使用除了第一帧之外的帧或帧来比较第一帧，将第二帧与由多个内插恢复处理生成的与第一帧近似的帧进行比较，计算接近于第一帧的每个帧的信噪比 通过处理第二帧作为信号，确定指示提供最高信噪比的内插恢复处理的索引号，以及使用编码参数来复用和发送索引号的第一帧。
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