公开(公告)号：US5732188A

公开(公告)日：1998-03-24

申请号：US612797

申请日：1996-03-11

申请人： Takehiro Moriya ,  Kazunori Mano ,  Satoshi Miki ,  Hitoshi Ohmura ,  Shigeaki Sasaki ,  Naoki Iwakami

发明人： Takehiro Moriya ,  Kazunori Mano ,  Satoshi Miki ,  Hitoshi Ohmura ,  Shigeaki Sasaki ,  Naoki Iwakami

IPC分类号： G10L19/00 ,  G10L19/04 ,  G10L19/06 ,  G10L19/12 ,  H03H17/02 ,  G10L9/04

CPC分类号： G10L19/06 ,  G10L25/24

摘要： In a CELP coding scheme, p-order LPC coefficients of an input signal are transformed into n-order LPC cepctrum coefficients c.sub.j (S.sub.2), which are modified into n-order modified LPC cepstrum coefficients c.sub.j ' (S.sub.3). Log power spectral envelopes of the input signal and a masking function suited thereto are calculated (FIGS. 3B, C), then they are subjected to inverse Fourier transform to obtain n-order LPC cepstrum coefficients, respectively, (FIGS. 3D, E), then the relationship between corresponding orders of the LPC cepstrum coefficients is calculated, and the modification in step S.sub.3 is carried out on the basis of the relationship. The modified coefficients c.sub.j are inversely transformed by the method of least squares into m-order LPC coefficients for use as filter coefficients of a perceptual weighting filter. This concept is applicable to a postfilter as well.

摘要翻译： 在CELP编码方案中，输入信号的p阶LPC系数被变换为n阶LPC位移系数cj（S2），其被修改为n阶修正LPC倒谱系数cj'（S3）。 计算输入信号的对数功率谱包络和适合其的屏蔽功能（图3B，C），然后对它们进行傅里叶逆变换以分别获得n阶LPC倒谱系数（图3D，E） 然后计算LPC倒谱系数的相应次序之间的关系，并且基于该关系来执行步骤S3中的修改。 经修正的系数cj通过最小二乘法的方法被逆变换为m阶LPC系数，用作感知加权滤波器的滤波器系数。 这个概念也适用于后置过滤器。

公开(公告)号：US5684920A

公开(公告)日：1997-11-04

申请号：US402660

申请日：1995-03-13

申请人： Naoki Iwakami ,  Takehiro Moriya ,  Satoshi Miki

发明人： Naoki Iwakami ,  Takehiro Moriya ,  Satoshi Miki

IPC分类号： G01R23/16 ,  G10L19/02 ,  G10L25/12 ,  G10L25/27 ,  G10L9/16

CPC分类号： G10L19/0212 ,  G10L19/0204 ,  G10L25/12 ,  G10L25/27

摘要： An input acoustic signal is subjected to modified discrete cosine transform processing to obtain its spectrum characteristics. Linear prediction coefficients are derived from the input acoustic signal in a linear prediction coding analysis part, and the prediction coefficients are subjected to Fourier transform in a spectrum envelope calculation part to obtain the envelope of the spectrum characteristics of the input acoustic signal. In a normalization part the spectrum characteristics are normalized by the envelope thereof to obtain residual coefficients. Another normalization part normalizes the residual coefficients by a residual-coefficients envelope predicted in a residual-coefficients envelope calculation part, thereby obtaining fine structure coefficients, which are vector-quantized in a quantization part. A de-normalization part de-normalizes the quantized fine structure coefficients. The residual-coefficients envelope calculation part uses the reproduced residual coefficients to predict the envelope of residual coefficients of the subsequent frame.

摘要翻译： 对输入声信号进行修正的离散余弦变换处理以获得其频谱特性。 线性预测系数从线性预测编码分析部中的输入声信号导出，并且在频谱包络计算部中对预测系数进行傅里叶变换，以获得输入声信号的频谱特性的包络。 在归一化部分中，光谱特性通过其包络进行归一化以获得残差系数。 另一个归一化部分通过在残差系数包络计算部分中预测的残差系数包络对残差系数进行归一化，从而获得在量化部分中矢量量化的精细结构系数。 去归一化部分使量化的精细结构系数解规范化。 剩余系数包络计算部分使用再现的残差系数来预测后续帧的残差系数的包络。

公开(公告)号：US5396576A

公开(公告)日：1995-03-07

申请号：US886013

申请日：1992-05-20

申请人： Satoshi Miki ,  Takehiro Moriya ,  Kazunori Mano ,  Hitoshi Ohmuro ,  Hirohito Suda

发明人： Satoshi Miki ,  Takehiro Moriya ,  Kazunori Mano ,  Hitoshi Ohmuro ,  Hirohito Suda

IPC分类号： G10L19/00 ,  G10L19/12 ,  G10L19/135 ,  G10L25/06 ,  G10L9/00

CPC分类号： G10L19/135 ,  G10L19/12 ,  G10L2019/0005 ,  G10L25/06

摘要： An excitation vector of the previous frame stored in an adaptive codebook is cut out with a selected pitch period. The excitation vector thus cut out is repeated until one frame is formed, by which a periodic component codevector is generated. An optimum pitch period is searched for so that distortion of a reconstructed speech obtained by exciting a linear predictive synthesis filter with the periodic component codevector is minimized. Thereafter, a random codevector selected from a random codebook is cut out with the optimum pitch period and is repeated until one frame is formed, by which a repetitious random codevector is generated. The random codebook is searched for a random codevector which minimizes the distortion of the reconstructed speech which is provided by exciting the synthesis filter with the repetitious random codevector.

摘要翻译： 存储在自适应码本中的先前帧的激励矢量以选定的音调周期被切除。 重复如此切出的激励矢量，直到形成一个帧，由此产生周期性分量码矢量。 搜索最佳音调周期，使得通过激励具有周期性分量码矢量的线性预测合成滤波器获得的重构语音的失真被最小化。 此后，以最佳音调周期切出从随机码本中选出的随机码矢量，并重复直到形成一个帧，由此产生重复的随机码矢量。 搜索随机码本的随机码矢量，其使通过用重复的随机码矢量激励合成滤波器提供的重构语音的失真最小化。

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：US06658382B1

公开(公告)日：2003-12-02

申请号：US09534297

申请日：2000-03-23

申请人： Naoki Iwakami ,  Takehiro Moriya ,  Akio Jin ,  Kazuaki Chikira ,  Takeshi Mori

发明人： Naoki Iwakami ,  Takehiro Moriya ,  Akio Jin ,  Kazuaki Chikira ,  Takeshi Mori

IPC分类号： G10L1902

CPC分类号： G10L19/0212

摘要： An input signal is time-frequency transformed, then the frequency-domain coefficients are divided into coefficient segments of about 100 Hz width to generate a sequence of coefficient segments, and the sequence of coefficient segments is split into subbands each consisting of plural coefficient segments. A threshold value is determined based on the intensity of each coefficient segment in each subband. The intensity of each coefficient segment is compared with the threshold value, and the coefficient segments are classified into low- and high-intensity groups. The coefficient segments are quantized for each group, or they are flattened respectively and then quantized through recombination.

摘要翻译： 输入信号被时间频率变换，频域系数被划分为大约100Hz宽度的系数段，以产生一系列系数段，并将系数段序列分成各个由多个系数段组成的子带。 基于每个子带中的每个系数段的强度来确定阈值。 将每个系数段的强度与阈值进行比较，将系数段分为低强度组和高强度组。 每个组对系数段进行量化，或者分别对其进行平坦化，然后通过重组进行量化。

公开(公告)号：US06345246B1

公开(公告)日：2002-02-05

申请号：US09018042

申请日：1998-02-03

申请人： Takehiro Moriya ,  Takeshi Mori ,  Kazunaga Ikeda ,  Naoki Iwakami

发明人： Takehiro Moriya ,  Takeshi Mori ,  Kazunaga Ikeda ,  Naoki Iwakami

IPC分类号： G10L1904

CPC分类号： G10L19/008

摘要： In multichannel acoustic signal coding and decoding, left- and right-channel signals are alternately interleaved for each sample to generate a one-dimensional signal sample sequence. The one-dimensional signal sample sequence is subjected to coding based on correlation. In coding, the left- and right-channel signals may preferably be interleaved after reducing an imbalance in power between input channels. In such an instance, a power imbalance is introduced between the decoded left- and right-channel signal sample sequences

摘要翻译： 在多声道声信号编码和解码中，左和右声道信号被交替地交替地用于每个采样以产生一维信号采样序列。 一维信号采样序列经过相关的编码。 在编码中，在减少输入通道之间的功率不平衡的情况下，左通道信号和右声道信号可优选地交错。 在这种情况下，在解码的左声道和右声道信号样本序列之间引入功率不平衡

公开(公告)号：US5651090A

公开(公告)日：1997-07-22

申请号：US433962

申请日：1995-05-04

申请人： Takehiro Moriya ,  Naoki Iwakami

发明人： Takehiro Moriya ,  Naoki Iwakami

IPC分类号： G06T9/00 ,  G10L19/00 ,  G10L19/008 ,  G10L19/02 ,  G10L19/038 ,  G10L19/12 ,  H04B1/66 ,  H04B14/04 ,  G10L3/02

CPC分类号： G10L19/038 ,  G06T9/008 ,  G10L19/008 ,  H04B1/667 ,  H04B14/046 ,  G10L19/00 ,  G10L19/0212 ,  G10L19/12

摘要： Power normalization parts calculate the average powers of input signals of a plurality of channels for each frame and divide the signals by the calculated average powers to generate normalized signals and, at the same time, generate weights corresponding to the normalization gains. The normalized signals of the plurality of channels are combined in a combining part into predetermined sequences and outputted therefrom as one or more interleaved signal vectors. The combining part combines the weights from the power normalization part into the same sequences of the normalized signal and outputs one or more interleaved weight vectors. In a vector quantization part the signal vectors are vector quantized by the interleaved weight vectors corresponding thereto, respectively, and quantization indexes and normalization indexes are outputted as results of coding.

摘要翻译： 功率归一化部件计算每个帧的多个信道的输入信号的平均功率，并将信号除以计算出的平均功率以产生归一化信号，并且同时产生对应于归一化增益的权重。 多个信道的归一化信号在组合部分中组合成预定的序列，并作为一个或多个交织的信号向量输出。 组合部分将来自功率归一化部分的权重组合成归一化信号的相同序列，并输出一个或多个交织权重向量。 在矢量量化部分中，信号矢量分别由与其对应的交织权重向量进行矢量量化，作为编码结果输出量化索引和归一化指标。

公开(公告)号：US06549147B1

公开(公告)日：2003-04-15

申请号：US09572789

申请日：2000-05-17

申请人： Takehiro Moriya ,  Naoki Iwakami ,  Akio Jin ,  Takeshi Mori ,  Kazuaki Chikira

发明人： Takehiro Moriya ,  Naoki Iwakami ,  Akio Jin ,  Takeshi Mori ,  Kazuaki Chikira

IPC分类号： H03M700

CPC分类号： H04N19/00 ,  H03M7/30 ,  H04N19/90

摘要： An object of the invention is to provide a method for compressing digital input signals at high compression efficiency and reproducing the input data perfectly. The method includes the steps of: converting a digital input signal in each frame to bitstreams according to a sign-magnitude format; deblocking the bitstreams into individual bits; joining each bit in a time sequence while retaining an identical chronological order of bits in all the frames; and reversibly encoding each bitstream obtained by joining the bits. And, the reversible decoding method includes the steps of: reversibly decoding a reversible code sequence in each frame; deblocking the bitstreams obtained by reversible decoding into individual bits; joining each bit in a time sequence while retaining an identical chronological order of bits in all the frames; and joining successive frames obtained by joining the bits.

摘要翻译： 本发明的目的是提供一种以高压缩效率压缩数字输入信号并且完美地再现输入数据的方法。 该方法包括以下步骤：根据符号幅度格式将每帧中的数字输入信号转换为比特流; 将比特流解块为单独的比特; 以时间顺序连接每个比特，同时保留所有帧中相同的时间顺序的比特; 并且可逆地编码通过连接比特获得的每个比特流。 并且，可逆解码方法包括以下步骤：对每帧中的可逆码序列进行可逆解码; 将通过可逆解码获得的比特流解块为单独的比特; 以时间顺序连接每个比特，同时保留所有帧中相同的时间顺序的比特; 并连接通过连接比特获得的连续帧。

公开(公告)号：US08576927B2

公开(公告)日：2013-11-05

申请号：US13061971

申请日：2009-10-07

申请人： Noboru Harada ,  Yutaka Kamamoto ,  Takehiro Moriya

发明人： Noboru Harada ,  Yutaka Kamamoto ,  Takehiro Moriya

IPC分类号： H04B14/04

CPC分类号： H03M7/50 ,  G10L19/0017

摘要： A frame formed of a plurality of code words encoded with an encoding mode in which two different types of code words are assigned one-to-one to two smallest quantization intervals is checked to determine whether it contains just the two types of code words assigned to the two smallest quantization intervals, and lossless encoding is applied to the frame containing just the two types of code words. A code obtained by this lossless encoding is decoded with a decoding method corresponding to the lossless encoding.

摘要翻译： 检查由多个编码模式形成的帧，其中两种不同类型的码字分配一对一到两个最小量化间隔的编码模式，以确定它是否仅包含分配给 将两个最小的量化间隔和无损编码应用于仅包含两种类型的码字的帧。 通过该无损编码获得的代码用对应于无损编码的解码方法解码。

公开(公告)号：US08341197B2

公开(公告)日：2012-12-25

申请号：US12518789

申请日：2007-12-28

申请人： Noboru Harada ,  Takehiro Moriya ,  Yutaka Kamamoto

发明人： Noboru Harada ,  Takehiro Moriya ,  Yutaka Kamamoto

IPC分类号： G06F12/00 ,  G06F15/16

CPC分类号： G06F17/30076 ,  G06F17/30123

摘要： In encoding for putting one or more files and/or one or more files in a folder into a single archive file, original hierarchy information and standard hierarchy information generated by converting characters in a special character coding in each file name to characters in a standard character coding are recorded in the archive file. If the character coding used in the original hierarchy information in the archive file cannot be used in the system environment used in decoding, a file name in the standard character coding is generated from the standard hierarchy information and is converted to a character coding that can be used in the system environment.

摘要翻译： 在将文件夹中的一个或多个文件和/或一个或多个文件放入单个归档文件的编码中，通过将每个文件名中的特殊字符编码中的字符转换为标准字符中的字符而生成的原始层级信息和标准层级信息 编码被记录在归档文件中。 如果归档文件中的原始层级信息中使用的字符编码不能在解码中使用的系统环境中使用，则从标准层级信息生成标准字符编码中的文件名，并将其转换成可以是 用于系统环境。