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1. 发明授权

US06484140B2 Apparatus and method for encoding a signal as well as apparatus and method for decoding signal 有权
标题翻译：用于编码信号的装置和方法以及用于解码信号的装置和方法
公开(公告)号：US06484140B2
公开(公告)日：2002-11-19
申请号：US09935881
申请日：2001-08-23
申请人： Jun Matsumoto , Masayuki Nishiguchi , Kenichi Makino
发明人： Jun Matsumoto , Masayuki Nishiguchi , Kenichi Makino
IPC分类号： G01L1914
CPC分类号： G10L19/0212 , G10L19/0204 , G10L19/09
摘要： An apparatus and a method for encoding an input signal on the time base through orthogonal transform involves removing the correlation of signal waveform based on parameters obtained by linear predictive coding (LPC) analysis and pitch analysis of the input signal on the time base prior to the orthogonal transform. A normalization circuit section removes the correlation of the signal waveform and takes out the residue by an LPC inverse filter and pitch inverse filter and sends the residue to an orthogonal transform circuit section. The LPC parameters and the pitch parameters are sent to a bit allocation calculation circuit. A coefficient quantization section quantizes the coefficients from the orthogonal transform circuit section according to the number of allocated bits from the bit allocation calculation section.
摘要翻译：通过正交变换在时基上对输入信号进行编码的装置和方法包括基于在线性预测编码（LPC）分析获得的参数和基于时间基准的输入信号的音调分析来消除信号波形的相关性正交变换。归一化电路部分去除信号波形的相关性，并通过LPC逆滤波器和音调反向滤波器取出残差，并将其发送到正交变换电路部分。 LPC参数和音调参数被发送到比特分配计算电路。系数量化部根据来自比特分配计算部的分配比特数，对来自正交变换电路部的系数进行量化。

2. 发明授权

US06353808B1 Apparatus and method for encoding a signal as well as apparatus and method for decoding a signal 有权
标题翻译：用于编码信号的装置和方法以及用于对信号进行解码的装置和方法
公开(公告)号：US06353808B1
公开(公告)日：2002-03-05
申请号：US09422250
申请日：1999-10-21
申请人： Jun Matsumoto , Masayuki Nishiguchi , Kenichi Makino
发明人： Jun Matsumoto , Masayuki Nishiguchi , Kenichi Makino
IPC分类号： G10L1904
CPC分类号： G10L19/0212 , G10L19/0204 , G10L19/09
摘要： An apparatus and a method for encoding an input signal on the time base through orthogonal transform involves removing the correlation of signal waveform on the basis of the parameters obtained by means of linear predictive coding (LPC) analysis and pitch analysis of the input signal on the time base prior to the orthogonal transform. The time base input signal from input terminal is sent to a normalization circuit section and a LPC analysis circuit. The normalization circuit section removes the correlation of the signal waveform and takes out the residue by an LPC inverse filter and pitch inverse filter and sends the residue to an orthogonal transform circuit section. The LPC parameters from the LPC analysis circuit and the pitch parameters from the pitch analysis circuit are sent to a bit allocation calculation circuit. A coefficient quantization section quantizes the coefficients from the orthogonal transform circuit section according to the number of allocated bits from the bit allocation calculation section.
摘要翻译：通过正交变换在时基上对输入信号进行编码的装置和方法包括基于通过线性预测编码（LPC）分析获得的参数和对输入信号的音调分析来去除信号波形的相关性在正交变换之前的时基。来自输入端子的时基输入信号被发送到归一化电路部分和LPC分析电路。归一化电路部分去除信号波形的相关性，并通过LPC逆滤波器和音调反向滤波器取出残差，并将其发送到正交变换电路部分。来自LPC分析电路的LPC参数和来自音调分析电路的音调参数被发送到比特分配计算电路。系数量化部根据来自比特分配计算部的分配比特数，对来自正交变换电路部的系数进行量化。

3. 发明授权

US06859816B2 Fast Fourier transform method and inverse fast Fourier transform method 失效
标题翻译：快速傅里叶变换法和快速傅里叶逆变换法
公开(公告)号：US06859816B2
公开(公告)日：2005-02-22
申请号：US09770963
申请日：2001-01-25
申请人： Kenichi Makino , Jun Matsumoto , Masayuki Nishiguchi
发明人： Kenichi Makino , Jun Matsumoto , Masayuki Nishiguchi
IPC分类号： G06F17/14
CPC分类号： G06F17/142
摘要： First, at step S1, i=0 is set. At step S2, data comprising M samples is fetched. At step S3, an M-point DFT is applied to the data fetched at step S2 above. At step S4, an obtained y(k) is multiplied by a twist coefficient w(i, k). The result is placed in y(k). At step S5, the value in y(k) is overwritten to an array x which contains original data. The above processing is repeated N/M times through steps S6 and S7 until all the input data is processed. At step S9, an FFT with N/M (=2n) points is performed within the range of 0≦k
摘要翻译：首先，在步骤S1，设定i = 0。在步骤S2，获取包含M个样本的数据。在步骤S3，将M点DFT应用于在上述步骤S2取出的数据。在步骤S4，将获得的y（k）乘以扭曲系数w（i，k）。结果放在y（k）中。在步骤S5，将y（k）中的值重写为包含原始数据的数组x。上述处理通过步骤S6和S7重复N / M次，直到所有输入数据被处理。在步骤S9，在0 <= k ）个点的FFT。最后，在步骤S12中进行排序。

4. 发明授权

US06681204B2 Apparatus and method for encoding a signal as well as apparatus and method for decoding a signal 失效
标题翻译：用于编码信号的装置和方法以及用于对信号进行解码的装置和方法
公开(公告)号：US06681204B2
公开(公告)日：2004-01-20
申请号：US09935931
申请日：2001-08-23
申请人： Jun Matsumoto , Masayuki Nishiguchi , Kenichi Makino
发明人： Jun Matsumoto , Masayuki Nishiguchi , Kenichi Makino
IPC分类号： G10L1902
CPC分类号： G10L19/0212 , G10L19/0204 , G10L19/09
摘要： An apparatus and a method for encoding an input signal on the time base through orthogonal transform involves removing the correlation of signal waveform on the basis of the parameters obtained by means of linear predictive coding (LPC) analysis and pitch analysis of the input signal on the time base prior to the orthogonal transform. The time base input signal from input terminal is sent to a normalization circuit section and a LPC analysis circuit. The normalization circuit section removes the correlation of the signal waveform and takes out the residue by an LPC inverse filter and pitch inverse filter and sends the residue to an orthogonal transform circuit section. The LPC parameters from the LPC analysis circuit and the pitch parameters from the pitch analysis circuit are sent to a bit allocation calculation circuit. A coefficient quantization section quantizes the coefficients from the orthogonal transform circuit section according to the number of allocated bits from the bit allocation calculation section.
摘要翻译：通过正交变换在时基上对输入信号进行编码的装置和方法包括基于通过线性预测编码（LPC）分析获得的参数和对输入信号的音调分析来去除信号波形的相关性在正交变换之前的时基。来自输入端子的时基输入信号被发送到归一化电路部分和LPC分析电路。归一化电路部分去除信号波形的相关性，并通过LPC逆滤波器和音调反向滤波器取出残差，并将其发送到正交变换电路部分。来自LPC分析电路的LPC参数和来自音调分析电路的音调参数被发送到比特分配计算电路。系数量化部根据来自比特分配计算部的分配比特数，对来自正交变换电路部的系数进行量化。

5. 发明授权

US07454330B1 Method and apparatus for speech encoding and decoding by sinusoidal analysis and waveform encoding with phase reproducibility 失效
标题翻译：通过正弦分析和具有相位再现性的波形编码进行语音编码和解码的方法和装置
公开(公告)号：US07454330B1
公开(公告)日：2008-11-18
申请号：US08736546
申请日：1996-10-24
申请人： Masayuki Nishiguchi , Kazuyuki Iijima , Jun Matsumoto , Shiro Omori
发明人： Masayuki Nishiguchi , Kazuyuki Iijima , Jun Matsumoto , Shiro Omori
IPC分类号： G10L19/14
CPC分类号： G10L19/0212 , G10L19/02 , G10L19/04 , G10L19/06 , G10L19/12 , G10L25/27 , G10L25/93
摘要： A speech encoding method and apparatus in which an input speech signal is divided in terms of blocks or frames as encoding units and encoded in terms of the encoding units, whereby explosive and fricative consonants can be impeccably reproduced, while there is an attenuation of the occurrence of foreign sounds being generated at a transient portion between voiced (V) and unvoiced (UV) portions, so that the speech with high clarity devoid of “stuffed” feeling may be produced. The encoding apparatus includes a first encoding unit for finding residuals of linear predictive coding (LPC) of an input speech signal for performing harmonic coding and a second encoding unit for encoding the input speech signal by waveform coding. The first encoding unit and the second encoding unit are used for encoding a voiced (V) portion and an unvoiced (UV) portion of the input signal, respectively. Code excited linear prediction (CELP) encoding employing vector quantization by a closed loop search of an optimum vector using an analysis-by-synthesis method is used for the second encoding unit. A corresponding decoding method and apparatus is also provided.
摘要翻译：一种语音编码方法和装置，其中输入语音信号以块或帧为单位编码，并以编码单位编码，由此可以无可挑剔地复制爆炸和摩擦辅音，同时存在衰减的发生在V（V）和无声（UV）部分之间的瞬态部分产生外来声音，从而可能产生具有高“透明度”感的语音。编码装置包括：第一编码单元，用于求出用于执行谐波编码的输入语音信号的线性预测编码（LPC）的残差;以及第二编码单元，用于通过波形编码对输入的语音信号进行编码。第一编码单元和第二编码单元分别用于对输入信号的有声（V）部分和无声（UV）部分进行编码。第二编码单元使用通过使用合成分析法的最佳向量的闭环搜索采用矢量量化的码激励线性预测（CELP）编码。还提供了相应的解码方法和装置。

6. 发明授权

US07373078B2 Audio signal processing method, audio signal processing apparatus, Hi-Fi video apparatus, digital video apparatus and 8 mm video apparatus 失效
公开(公告)号：US07373078B2
公开(公告)日：2008-05-13
申请号：US11112340
申请日：2005-04-22
申请人： Mototsugu Abe , Akira Inoue , Jun Matsumoto , Koji Suginuma , Masayuki Nishiguchi
发明人： Mototsugu Abe , Akira Inoue , Jun Matsumoto , Koji Suginuma , Masayuki Nishiguchi
IPC分类号： H04N7/00 , H04N7/64
CPC分类号： G11B15/026 , G11B5/0086 , G11B5/531 , G11B15/125 , G11B15/1875 , G11B20/10527 , G11B20/1201 , G11B20/18 , G11B20/1876 , G11B2020/10546 , H04N5/78213 , H04N5/78263 , H04N5/783 , H04N9/8063
摘要： An audio signal processing method for repairing an anomalous state such as noise, a discontinuity, and a break of sound, comprising detecting the anomalous state of an audio signal, deleting the audio signal in the anomalous segment, deducing the correct audio signal by referring to the waveform of the audio signal before and after the deleted segment, generating a repair signal for repairing the signal in the deleted segment based on the deduced result, inserting the repair signal into the deleted segment, and connecting it to the audio signal before and after the deleted segment.

7. 发明授权

US07248789B2 Audio signal processing method, audio signal processing apparatus, Hi-Fi video apparatus, digital video apparatus and 8 mm video apparatus 有权
公开(公告)号：US07248789B2
公开(公告)日：2007-07-24
申请号：US11112398
申请日：2005-04-22
申请人： Mototsugu Abe , Akira Inoue , Jun Matsumoto , Koji Suginuma , Masayuki Nishiguchi
发明人： Mototsugu Abe , Akira Inoue , Jun Matsumoto , Koji Suginuma , Masayuki Nishiguchi
IPC分类号： H04N7/00 , H04N7/64
CPC分类号： G11B15/026 , G11B5/0086 , G11B5/531 , G11B15/125 , G11B15/1875 , G11B20/10527 , G11B20/1201 , G11B20/18 , G11B20/1876 , G11B2020/10546 , H04N5/78213 , H04N5/78263 , H04N5/783 , H04N9/8063
摘要： An audio signal processing method for repairing an anomalous state such as noise, a discontinuity, and a break of sound, comprising detecting the anomalous state of an audio signal, deleting the audio signal in the anomalous segment, deducing the correct audio signal by referring to the waveform of the audio signal before and after the deleted segment, generating a repair signal for repairing the signal in the deleted segment based on the deduced result, inserting the repair signal into the deleted segment, and connecting it to the audio signal before and after the deleted segment.

8. 发明授权

US07242855B2 Audio signal processing method, audio signal processing apparatus, Hi-Fi video apparatus, digital video apparatus and 8 mm video apparatus 有权
标题翻译：音频信号处理方法，音频信号处理装置，Hi-Fi视频装置，数字视频装置和8mm视频装置
公开(公告)号：US07242855B2
公开(公告)日：2007-07-10
申请号：US11112397
申请日：2005-04-22
申请人： Mototsugu Abe , Akira Inoue , Jun Matsumoto , Koji Suginuma , Masayuki Nishiguchi
发明人： Mototsugu Abe , Akira Inoue , Jun Matsumoto , Koji Suginuma , Masayuki Nishiguchi
IPC分类号： H04N7/00 , H04N7/64
CPC分类号： G11B15/026 , G11B5/0086 , G11B5/531 , G11B15/125 , G11B15/1875 , G11B20/10527 , G11B20/1201 , G11B20/18 , G11B20/1876 , G11B2020/10546 , H04N5/78213 , H04N5/78263 , H04N5/783 , H04N9/8063
摘要： An audio signal processing method for repairing an anomalous state such as noise, a discontinuity, and a break of sound, comprising detecting the anomalous state of an audio signal, deleting the audio signal in the anomalous segment, deducing the correct audio signal by referring to the waveform of the audio signal before and after the deleted segment, generating a repair signal for repairing the signal in the deleted segment based on the deduced result, inserting the repair signal into the deleted segment, and connecting it to the audio signal before and after the deleted segment.
摘要翻译：一种音频信号处理方法，用于修复诸如噪声，不连续性和中断声音的异常状态，包括检测音频信号的异常状态，删除异常段中的音频信号，通过参考以下内容推导出正确的音频信号在删除段之前和之后的音频信号的波形，基于推导结果生成用于修复被删除段中的信号的修复信号，将修复信号插入到删除的段中，并将其连接到前后的音频信号已删除的段。

9. 发明授权

US5809455A Method and device for discriminating voiced and unvoiced sounds 失效
标题翻译：用于辨别有声和无声的声音的方法和装置
公开(公告)号：US5809455A
公开(公告)日：1998-09-15
申请号：US753347
申请日：1996-11-25
申请人： Masayuki Nishiguchi , Jun Matsumoto
发明人： Masayuki Nishiguchi , Jun Matsumoto
IPC分类号： G10L11/02 , G06F17/18 , G10L11/06 , G10L15/02 , G10L15/04 , G10L3/02
CPC分类号： G10L25/93 , G10L2025/783 , G10L2025/932
摘要： A method and a device for discriminating a voiced sound from an unvoiced sound or background noise in speech signals are disclosed. Each block or frame of input speech signals is divided into plural sub-blocks and the standard deviation, effective value or the peak value is detected in a detection unit for detecting statistical characteristics from one sub-block to another. A bias detection unit detects a bias on the time scale of the standard deviation, effective value or the peak value to decide whether the speech signals are voiced or unvoiced from one block to another.
摘要翻译：公开了一种在语音信号中识别有声声音与无声或背景噪声的方法和装置。输入语音信号的每个块或帧被分成多个子块，并且在用于检测从一个子块到另一个子块的统计特性的检测单元中检测标准偏差，有效值或峰值。偏置检测单元检测标准偏差，有效值或峰值的时间标度上的偏差，以确定语音信号是从一个块到另一个是浊音还是清音。

10. 发明授权

US5749065A Speech encoding method, speech decoding method and speech encoding/decoding method 失效
标题翻译：语音编码方法，语音解码方式和语音编码/解码方法
公开(公告)号：US5749065A
公开(公告)日：1998-05-05
申请号：US518298
申请日：1995-08-23
申请人： Masayuki Nishiguchi , Jun Matsumoto
发明人： Masayuki Nishiguchi , Jun Matsumoto
IPC分类号： G10L19/08 , G10L11/06 , G10L19/00 , G10L19/04 , G10L19/14 , G10L3/02 , G10L9/00
CPC分类号： G10L19/18 , G10L25/93
摘要： A speech encoding/decoding method calculates a short-term prediction error of an input speech signal that is divided on a time axis into blocks, represents the short-term prediction residue by a synthesized sine wave and a noise and encodes a frequency spectrum of each of the synthesized sine wave and the noise to encode the speech signal. The speech encoding/decoding method decodes the speech signal on a block basis and finds a short-term prediction residue waveform by sine wave synthesis and noise synthesis of the encoded speech signal. The speech encoding/decoding method then synthesizes the time-axis waveform signal based on the short-term prediction residue waveform of the encoded speech signal.
摘要翻译：语音编码/解码方法将在时间轴上划分的输入语音信号的短期预测误差计算成块，通过合成正弦波和噪声表示短期预测残差，并编码每个的合成正弦波和噪声来编码语音信号。语音编码/解码方法以块为基础对语音信号进行解码，并通过编码语音信号的正弦波合成和噪声合成找到短期预测残差波形。然后，语音编码/解码方法基于编码语音信号的短期预测残差波形来合成时间轴波形信号。

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