公开(公告)号：USRE38279E1

公开(公告)日：2003-10-21

申请号：US09691862

申请日：2000-10-19

申请人： Akitoshi Kataoka ,  Jotaro Ikedo

发明人： Akitoshi Kataoka ,  Jotaro Ikedo

IPC分类号： G01G918

CPC分类号： H03M7/02 ,  G06T9/008 ,  G10L19/00 ,  G10L19/005 ,  G10L2019/0005 ,  G10L2019/0013 ,  G10L2019/0014 ,  H03M7/3082

摘要： Representative vectors Z1i and Z2j are selected from code-books codebooks CB1 and CB1 CB2, respectively, and multiplied by weighting coefficient vectors w1 and w2 of the same number of dimensions as those of the representative vectors, whereby weighted representative vectors Z1iw1 and Z2jw2 are generated. These weighted representative vectors are vector combined into a combined vector yij, and a combination of the representative vectors is selected by a control part in such a manner as to minimize the distance between the combined vector yij and an input vector X. The weighting coefficient vectors w1 and w2 each have a maximum component in a different dimension and are selected so that the sum of diagonal matrixes W1 and W2 using components of the weighting coefficient vectors as their diagonal elements becomes a constant multiple of the unit matrix.

公开(公告)号：US5825311A

公开(公告)日：1998-10-20

申请号：US793133

申请日：1997-04-23

申请人： Akitoshi Kataoka ,  Jotaro Ikedo

发明人： Akitoshi Kataoka ,  Jotaro Ikedo

IPC分类号： G10L15/02 ,  G06T9/00 ,  G10L19/00 ,  G10L19/12 ,  H03M7/02 ,  H03M7/30 ,  G10L9/18

CPC分类号： H03M7/02 ,  G06T9/008 ,  G10L19/00 ,  G10L19/005 ,  H03M7/3082

摘要： Representative vectors z.sub.1i and z.sub.2j are selected from codebooks CB1 and CB1, respetively, and multiplied by weighting coefficient vectors w.sub.1 and w.sub.2 of the same number of dimensions as those of the representative vectors, whereby weighted representative vectors z.sub.1i w.sub.1 and z.sub.2j w.sub.2 are generated. These weighted representative vectors are vector combined into a combined vector y.sub.ij, and a combination of the representative vectors is selected by a control part in such a manner as to minimize the distance between the combined vector y.sub.ij and an input vector X. The weighting coefficient vectors w.sub.1 and w.sub.2 each have a maximum component in a different dimension and are selected so that the sum of diagonal matrixes W.sub.1 and W.sub.2 using components of the weighting coefficient vectors as their diagonal elements becomes a constant multiple of the unit matrix.

摘要翻译： PCT No.PCT / JP95 / 01989 Sec。 371日期1997年4月23日 102（e）日期1997年4月23日PCT提交1995年9月29日PCT公布。 公开号WO96 / 11468 PCT 日期1996年4月18日代表矢量z1i和z2j从码本CB1和CB1中分别选择，并乘以与代表矢量相同维数的加权系数向量w1和w2，由此加权代表向量z1iw1和z2jw2是 生成。 这些加权代表矢量是矢量组合成组合矢量yij，并且控制部分选择代表矢量的组合，以使组合矢量yij和输入矢量X之间的距离最小化。加权系数矢量 w1和w2各自具有不同维度的最大分量，并且被选择为使得使用加权系数向量的分量作为对角元素的对角矩阵W1和W2的和变为单位矩阵的常数倍数。

公开(公告)号：US20090103517A1

公开(公告)日：2009-04-23

申请号：US10584833

申请日：2005-05-10

申请人： Hitoshi Ohmuro ,  Takeshi Mori ,  Yusuke Hiwasaki ,  Akitoshi Kataoka

发明人： Hitoshi Ohmuro ,  Takeshi Mori ,  Yusuke Hiwasaki ,  Akitoshi Kataoka

IPC分类号： H04L12/66

CPC分类号： G10L19/005 ,  G10L19/167 ,  H04L65/602 ,  H04L65/607 ,  H04L65/80

摘要： When acoustic signal packets are communicated over an IP communication network, data corresponding to an acoustic signal (acoustic signal corresponding data) has been included and transmitted in a packet different from a packet containing the acoustic signal. However, conventionally, a packet in which the acoustic signal corresponding data is to be included must be determined beforehand and cannot dynamically be changed.According to the present invention, the amount of delay of acoustic signal corresponding data with respect to an acoustic signal is contained in an acoustic signal packet as delay amount control information. Furthermore, the conditions of a communication network are detected from the number of packets lost in a burst loss or jitters and the number of the packets to be stored and the amount of delay at the receiving end are thereby determined.

摘要翻译： 当通过IP通信网络传送声信号分组时，已经包含与声信号（声信号对应数据）对应的数据，并且在与包含声信号的分组不同的分组中发送。 然而，传统上，必须预先确定其中要包括声信号对应数据的分组，并且不能动态地改变。 根据本发明，相对于声信号的声信号对应数据的延迟量被包含在声信号分组中作为延迟量控制信息。 此外，从突发丢失或抖动中丢失的分组的数量和要存储的分组的数量和接收端的延迟量来确定通信网络的条件。

公开(公告)号：US20070177620A1

公开(公告)日：2007-08-02

申请号：US10591183

申请日：2005-05-25

申请人： Hitoshi Ohmuro ,  Takeshi Mori ,  Yusuke Hiwasaki ,  Akitoshi Kataoka

发明人： Hitoshi Ohmuro ,  Takeshi Mori ,  Yusuke Hiwasaki ,  Akitoshi Kataoka

IPC分类号： H04L12/56 ,  H04J3/06

CPC分类号： G10L19/005 ,  G10L21/04 ,  H04J3/0632

摘要： The present invention prevents a receiving buffer from becoming empty by: storing received packets in the receiving buffer; detecting the largest arrival delay jitter of the packets and the buffer level of the receiving buffer by a state detecting part; obtaining an optimum buffer level for the largest delay jitter using a predetermined table by a control part; determining, based on the detected buffer level and the optimum buffer level, the level of urgency about the need to adjust the buffer level; expanding or reducing the waveform of a decoded audio data stream of the current frame decoded from a packet read out of the receiving buffer by a consumption adjusting part to adjust the consumption of reproduction frames on the basis of the urgency level, the detected buffer level, and the optimum buffer level.

摘要翻译： 本发明通过以下操作来防止接收缓冲器变空：将接收到的分组存储在接收缓冲器中; 通过状态检测部分检测分组的最大到达延迟抖动和接收缓冲器的缓冲器电平; 通过控制部分使用预定表获得用于最大延迟抖动的最佳缓冲器电平; 基于检测到的缓冲器级别和最佳缓冲器级别确定关于需要调整缓冲器级别的紧急程度; 扩展或减少由消耗调整部分从接收缓冲器读出的分组中解码的当前帧解码音频数据流的波形，以根据紧急程度调整再现帧的消耗，检测缓冲器电平， 和最佳缓冲区级别。

公开(公告)号：US08724734B2

公开(公告)日：2014-05-13

申请号：US12812549

申请日：2009-01-23

申请人： Shigeaki Sasaki ,  Takeshi Mori ,  Hitoshi Ohmuro ,  Yusuke Hiwasaki ,  Akitoshi Kataoka ,  Kimitaka Tsutsumi

发明人： Shigeaki Sasaki ,  Takeshi Mori ,  Hitoshi Ohmuro ,  Yusuke Hiwasaki ,  Akitoshi Kataoka ,  Kimitaka Tsutsumi

IPC分类号： H04L27/00

CPC分类号： H03M7/3082 ,  G10L19/00 ,  H04N19/194 ,  H04N19/85 ,  H04N19/94

摘要： A coding method with a small error is provided. In the coding method of the present invention, a normalization value obtained from an input signal is corrected for an error calculated from an input and output in vector quantization and is then quantized. The coding method includes a normalization stage of normalizing the input signal in accordance with the normalization value of the input signal, calculated in each frame; a dividing stage of dividing the normalized frame into divided input signal sequences in accordance with a predetermined rule; a vector quantization stage of applying vector quantization to the divided input signal sequences to generate a vector quantization index; and a normalization value correction stage of correcting the normalization value of the input signal for the error obtained from the input and output in the vector quantization stage.

摘要翻译： 提供了一个小错误的编码方法。 在本发明的编码方法中，对于从矢量量化中的输入和输出计算的误差校正从输入信号获得的归一化值，然后对其进行量化。 编码方法包括根据在每帧中计算的输入信号的归一化值对输入信号进行归一化的归一化阶段; 根据预定规则将归一化帧划分成分割输入信号序列的分割级; 将矢量量化应用于分割输入信号序列以生成矢量量化索引的矢量量化级; 以及归一化值校正级，用于校正从矢量量化级中的输入和输出获得的误差的输入信号的归一化值。

公开(公告)号：US07711554B2

公开(公告)日：2010-05-04

申请号：US10580195

申请日：2005-05-10

申请人： Takeshi Mori ,  Hitoshi Ohmuro ,  Yusuke Hiwasaki ,  Akitoshi Kataoka

发明人： Takeshi Mori ,  Hitoshi Ohmuro ,  Yusuke Hiwasaki ,  Akitoshi Kataoka

IPC分类号： G10L19/14 ,  G10L21/02 ,  G10L21/04

CPC分类号： G10L19/005 ,  G10L19/167 ,  G10L25/69

摘要： Input speech is coded in an encoder (11), the coded speech is decoded in a decoder (12), compensatory speech which compensates the speech of the current frame is generated in a compensatory speech generating part (20) by using past decoded speech, the quality of the compensatory speech is evaluated by using the input speech and the compensatory speech and a duplication level is generated the value of which increases incrementally with decreasing speech quality evaluation value in a speech quality evaluating part (40), and as many identical packets as the number specified by the duplication level is generated for the coded speech in a packet generating part (15), and the packets are transmitted, thereby reducing the possibility that packet loss will occur at the receiving end.

摘要翻译： 输入语音在编码器（11）中编码，编码语音在解码器（12）中解码，补偿语音通过使用过去的解码语音在补偿语音产生部分（20）中产生当前帧的语音， 通过使用输入语音和补偿语音来评估补偿语音的质量，并且产生其语音质量评估部分（40）中随着语音质量评估值的降低逐渐增加的复制度，以及许多相同的分组 由于在分组生成部（15）中为编码语音生成由复制级别指定的数字，并且发送分组，从而降低了在接收端发生分组丢失的可能性。

公开(公告)号：US5787391A

公开(公告)日：1998-07-28

申请号：US658303

申请日：1996-06-05

申请人： Takehiro Moriya ,  Akitoshi Kataoka ,  Kazunori Mano ,  Satoshi Miki ,  Hitoshi Omuro ,  Shinji Hayashi

发明人： Takehiro Moriya ,  Akitoshi Kataoka ,  Kazunori Mano ,  Satoshi Miki ,  Hitoshi Omuro ,  Shinji Hayashi

IPC分类号： G10L19/005 ,  G10L19/06 ,  G10L19/07 ,  G10L19/08 ,  G10L19/083 ,  G10L19/12 ,  G10L19/135 ,  G10L3/02

CPC分类号： G10L19/08 ,  G10L19/005 ,  G10L19/06 ,  G10L19/07 ,  G10L19/083 ,  G10L19/12 ,  G10L19/135

摘要： In a speech coding method of the present invention, initially, a plurality of samples of speech data are analyzed by a linear prediction analysis and thereby prediction coefficients are calculated. Then, the prediction coefficients are quantized, and the quantized prediction coefficients are set in a synthesis filter. Moreover, a pitch period vector is selected from an adaptive codebook in which a plurality of pitch period vectors are stored, and the selected pitch period vector is multiplied by a first gain which is obtained, at the same time, with a second gain. In addition, a noise waveform vector is selected from a random codebook in which a plurality of the noise waveform vectors are stored, and is multiplied by a predicted gain and the second gain. Then, the speech vector is synthesized by exciting the synthesis filter with the pitch period vector multiplied by the first gain, and with the noise waveform vector multiplied by the predicted gain and the second gain. Consequently, speech data comprising a plurality of samples are coded as a unit of a frame operation. Furthermore, the predicted gain multiplied by the noise waveform vector which is selected in a subsequent frame operation, is predicted based on the current noise waveform vector which is multiplied by the predicted gain and the second gain at the current frame operation, and also the previous waveform vector which is multiplied by the predicted gain and the second gain in the previous frame operation.

摘要翻译： 在本发明的语音编码方法中，首先，通过线性预测分析来分析多个语音数据样本，从而计算出预测系数。 然后，量化预测系数，并将量化的预测系数设置在合成滤波器中。 此外，从存储多个音调周期矢量的自适应码本中选择音调周期矢量，并且将所选择的音调周期矢量乘以与第二增益同时获得的第一增益。 此外，从存储多个噪声波形向量的随机码本中选择噪声波形向量，并将其乘以预测的增益和第二增益。 然后，通过利用乘以第一增益的音调周期矢量并且噪声波形向量乘以预测增益和第二增益来激励合成滤波器来合成语音向量。 因此，包括多个样本的语音数据被编码为帧操作的单位。 此外，基于在当前帧操作中乘以预测增益和第二增益的当前噪声波形向量来预测在后续帧操作中选择的噪声波形向量的预测增益，以及前一帧 在前一帧操作中乘以预测增益和第二增益的波形向量。

公开(公告)号：US08320391B2

公开(公告)日：2012-11-27

申请号：US10584833

申请日：2005-05-10

申请人： Hitoshi Ohmuro ,  Takeshi Mori ,  Yusuke Hiwasaki ,  Akitoshi Kataoka

发明人： Hitoshi Ohmuro ,  Takeshi Mori ,  Yusuke Hiwasaki ,  Akitoshi Kataoka

IPC分类号： H04L12/56 ,  G10L19/12 ,  G10L19/00

CPC分类号： G10L19/005 ,  G10L19/167 ,  H04L65/602 ,  H04L65/607 ,  H04L65/80

摘要： When acoustic signal packets are communicated over an IP communication network, data corresponding to an acoustic signal (acoustic signal corresponding data) has been included and transmitted in a packet different from a packet containing the acoustic signal. However, conventionally, a packet in which the acoustic signal corresponding data is to be included must be determined beforehand and cannot dynamically be changed.According to the present invention, the amount of delay of acoustic signal corresponding data with respect to an acoustic signal is contained in an acoustic signal packet as delay amount control information. Furthermore, the conditions of a communication network are detected from the number of packets lost in a burst loss or jitters and the number of the packets to be stored and the amount of delay at the receiving end are thereby determined.

摘要翻译： 当通过IP通信网络传送声信号分组时，已经包含与声信号（声信号对应数据）对应的数据，并且在与包含声信号的分组不同的分组中发送。 然而，传统上，必须预先确定其中要包括声信号对应数据的分组，并且不能动态地改变。 根据本发明，相对于声信号的声信号对应数据的延迟量被包含在声信号分组中作为延迟量控制信息。 此外，从突发丢失或抖动中丢失的分组的数量和要存储的分组的数量和接收端的延迟量来确定通信网络的条件。

公开(公告)号：US20110044405A1

公开(公告)日：2011-02-24

申请号：US12812549

申请日：2009-01-23

申请人： Shigeaki Sasaki ,  Takeshi Mori ,  Hitoshi Ohmuro ,  Yusuke Hiwasaki ,  Akitoshi Kataoka ,  Kimitaka Tsutsumi

发明人： Shigeaki Sasaki ,  Takeshi Mori ,  Hitoshi Ohmuro ,  Yusuke Hiwasaki ,  Akitoshi Kataoka ,  Kimitaka Tsutsumi

IPC分类号： H04L25/49

CPC分类号： H03M7/3082 ,  G10L19/00 ,  H04N19/194 ,  H04N19/85 ,  H04N19/94

摘要： A coding method with a small error is provided. In the coding method of the present invention, a normalization value obtained from an input signal is corrected for an error calculated from an input and output in vector quantization and is then quantized. The coding method includes a normalization stage of normalizing the input signal in accordance with the normalization value of the input signal, calculated in each frame; a dividing stage of dividing the normalized frame into divided input signal sequences in accordance with a predetermined rule; a vector quantization stage of applying vector quantization to the divided input signal sequences to generate a vector quantization index; and a normalization value correction stage of correcting the normalization value of the input signal for the error obtained from the input and output in the vector quantization stage.

摘要翻译： 提供了一个小错误的编码方法。 在本发明的编码方法中，对于从矢量量化中的输入和输出计算的误差校正从输入信号获得的归一化值，然后对其进行量化。 编码方法包括根据在每帧中计算的输入信号的归一化值对输入信号进行归一化的归一化阶段; 根据预定规则将归一化帧划分成分割输入信号序列的分割级; 将矢量量化应用于分割输入信号序列以生成矢量量化索引的矢量量化级; 以及归一化值校正级，用于校正从矢量量化级中的输入和输出获得的误差的输入信号的归一化值。

公开(公告)号：US07710982B2

公开(公告)日：2010-05-04

申请号：US10591183

申请日：2005-05-25

申请人： Hitoshi Ohmuro ,  Takeshi Mori ,  Yusuke Hiwasaki ,  Akitoshi Kataoka

发明人： Hitoshi Ohmuro ,  Takeshi Mori ,  Yusuke Hiwasaki ,  Akitoshi Kataoka

IPC分类号： H04L12/28

CPC分类号： G10L19/005 ,  G10L21/04 ,  H04J3/0632

摘要： The present invention prevents a receiving buffer from becoming empty by: storing received packets in the receiving buffer; detecting the largest arrival delay jitter of the packets and the buffer level of the receiving buffer by a state detecting part; obtaining an optimum buffer level for the largest delay jitter using a predetermined table by a control part; determining, based on the detected buffer level and the optimum buffer level, the level of urgency about the need to adjust the buffer level; expanding or reducing the waveform of a decoded audio data stream of the current frame decoded from a packet read out of the receiving buffer by a consumption adjusting part to adjust the consumption of reproduction frames on the basis of the urgency level, the detected buffer level, and the optimum buffer level.

摘要翻译： 本发明通过以下方法防止接收缓冲器变空：将接收到的数据包存储在接收缓冲器中; 通过状态检测部分检测分组的最大到达延迟抖动和接收缓冲器的缓冲器电平; 通过控制部分使用预定表获得用于最大延迟抖动的最佳缓冲器电平; 基于检测到的缓冲器级别和最佳缓冲器级别确定关于需要调整缓冲器级别的紧急程度; 扩展或减少由消耗调整部分从接收缓冲器读出的分组中解码的当前帧解码音频数据流的波形，以根据紧急程度调整再现帧的消耗，检测缓冲器电平， 和最佳缓冲区级别。