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1. 发明授权

US07728741B2 Code conversion device, code conversion method used for the same and program thereof 失效
标题翻译：代码转换装置，代码转换方法及其程序
公开(公告)号：US07728741B2
公开(公告)日：2010-06-01
申请号：US12158220
申请日：2006-12-19
申请人： Atsushi Murashima
发明人： Atsushi Murashima
IPC分类号： H03M5/00
CPC分类号： G10L19/173 , G10L19/005 , G10L19/167 , H04N19/40 , H04N19/61 , H04N19/895
摘要： Provided is a code conversion device that is capable of converting codes even if an input code sequence is invalid, and is able to reduce the amount of processing. When a first code sequence is input, the code conversion device generates a decoded signal by decoding the codes of normal frames of the first code sequence at Step S1, stores and holds the decoded signal at Step S2, generates a signal corresponding to an invalid frame by interpolation with the decoded signal that is stored and held, at Step S3. Subsequently, the code conversion device generates codes corresponding to the invalid frame by encoding the generated signal at Step S4, and makes the normal frames of the first code sequence without conversion be the frames of the second code sequence while making the generated codes be the frame of the second code sequence, in place of the codes of the invalid frame, at Step S5.
摘要翻译：提供了即使输入代码序列无效也能够转换代码并且能够减少处理量的代码转换装置。当输入第一代码序列时，代码转换装置通过在步骤S1解码第一代码序列的正常帧的代码来产生解码信号，在步骤S2存储和保持解码信号，产生对应于无效帧的信号通过对存储和保持的解码信号进行内插，在步骤S3中。随后，代码转换装置通过在步骤S4对所生成的信号进行编码来生成与无效帧对应的代码，并且使第一代码序列的正常帧不转换为第二代码序列的帧，同时使生成的代码为帧的第二代码序列代替无效帧的代码。

2. 发明授权

US07698135B2 Voice detecting method and apparatus using a long-time average of the time variation of speech features, and medium thereof 失效
标题翻译：使用语音特征的时间变化的长时间平均的语音检测方法和装置及其介质
公开(公告)号：US07698135B2
公开(公告)日：2010-04-13
申请号：US11501958
申请日：2006-08-10
申请人： Atsushi Murashima
发明人： Atsushi Murashima
IPC分类号： G10L15/20
CPC分类号： G10L25/78
摘要： A first filter (2061 in FIG. 1) calculates a long-time average of first change quantities based on a difference between a line spectral frequency of an input voice signal and a long-time average thereof. A second filter (2062 in FIG. 1) calculates a long-time average of second change quantities based on a difference between a whole band energy of the input voice signal and a long-time average thereof. A third filter (2063 in FIG. 1) calculates a long-time average of third change quantities based on a difference between a low band energy of the input voice signal and a long-time average thereof. A fourth filter (2064 in FIG. 1) calculates a long-time average of fourth change quantities based on a difference between a zero cross number of the input voice signal and a long-time average thereof. A voice/non-voice determining circuit (1040 in FIG. 1) discriminates a voice section from a non-voice section in the voice signal using the long-time average of the above-described first change quantities, the long-time average of the above-described second change quantities, the long-time average of the above-described third change quantities, and the long-time average of the above-described fourth change quantities.
摘要翻译：第一滤波器（图1中的2061）基于输入语音信号的线频谱频率与其长时间平均值之差来计算第一变化量的长时间平均值。第二滤波器（图1中的2062）基于输入语音信号的整个频带能量与其长时间平均值之差来计算第二变化量的长时间平均值。第三滤波器（图1中的2063）基于输入语音信号的低频带能量与其长时间平均值之差来计算第三变化量的长时间平均值。第四滤波器（图1中的2064）基于输入语音信号的零交叉数与其长时间平均值之差来计算第四变化量的长时间平均值。语音/非语音确定电路（图1中的1040）使用上述第一变化量的长时间平均值来区分语音信号中的语音部分与语音信号中的非语音部分，长时间平均值上述第二变化量，上述第三变化量的长时间平均值和上述第四变化量的长时间平均值。

3. 发明授权

US07117150B2 Voice detecting method and apparatus using a long-time average of the time variation of speech features, and medium thereof 失效
标题翻译：使用语音特征的时间变化的长时间平均的语音检测方法和装置及其介质
公开(公告)号：US07117150B2
公开(公告)日：2006-10-03
申请号：US09871368
申请日：2001-05-31
申请人： Atsushi Murashima
发明人： Atsushi Murashima
IPC分类号： G10L11/06
CPC分类号： G10L25/78
摘要： A first filter (2061 in FIG. 1) calculates a long-time average of first change quantities based on a difference between a line spectral frequency of an input voice signal and a long-time average thereof. A second filter (2062 in FIG. 1) calculates a long-time average of second change quantities based on a difference between a whole band energy of the input voice signal and a long-time average thereof. A third filter (2063 in FIG. 1) calculates a long-time average of third change quantities based on a difference between a low band energy of the input voice signal and a long-time average thereof. A fourth filter (2064 in FIG. 1) calculates a long-time average of fourth change quantities based on a difference between a zero cross number of the input voice signal and a long-time average thereof. A voice/non-voice determining circuit (1040 in FIG. 1) discriminates a voice section from a non-voice section in the voice signal using the long-time average of the above-described first change quantities, the long-time average of the above-described second change quantities, the long-time average of the above-described third change quantities, and the long-time average of the above-described fourth change quantities.
摘要翻译：第一滤波器（图1中的2061）基于输入语音信号的线频谱频率与其长时间平均值之差来计算第一变化量的长时间平均值。第二滤波器（图1中的2062）基于输入语音信号的整个频带能量与其长时间平均值之差来计算第二变化量的长时间平均值。第三滤波器（图1中的2063）基于输入语音信号的低频带能量与其长时间平均值之差来计算第三变化量的长时间平均值。第四滤波器（图1中的2064）基于输入语音信号的零交叉数与其长时间平均值之差来计算第四变化量的长时间平均值。语音/非语音确定电路（图1中的1040）使用上述第一变化量的长时间平均值来区分语音信号中的语音部分与语音信号中的非语音部分，长时间平均值上述第二变化量，上述第三变化量的长时间平均值和上述第四变化量的长时间平均值。

4. 发明授权

US06910009B1 Speech signal decoding method and apparatus, speech signal encoding/decoding method and apparatus, and program product therefor 有权
标题翻译：语音信号解码方法及装置，语音信号编码/解码方法及装置及其程序产品
公开(公告)号：US06910009B1
公开(公告)日：2005-06-21
申请号：US09699435
申请日：2000-10-31
申请人： Atsushi Murashima
发明人： Atsushi Murashima
IPC分类号： G10L19/06 , G10L19/00 , G10L19/08 , G10L19/12 , H03M7/30 , G10L19/14
CPC分类号： G10L19/083 , G10L21/0364 , G10L2019/0012
摘要： The quality of reconstructed speech on which background noise is superimposed is improved in a speech signal decoding apparatus for generating a speech signal by driving a filter, which is constituted by linear prediction coefficients, by an excitation signal. A smoothing circuit smoothes sound source gain in a noise segment using sound source gain that was obtained in the past. A smoothing-quantity limiting circuit calculates an amount of fluctuation represented by dividing, by the sound source gain, the absolute value of the difference between the sound source gain and the sound source gain that has been smoothed, and limits the value of the smoothed gain in such a manner that the amount of fluctuation will not exceed a certain threshold value.
摘要翻译：在用于通过用激励信号驱动由线性预测系数构成的滤波器来产生语音信号的语音信号解码装置中，重叠了背景噪声的重构语音的质量得到改善。平滑电路使用过去获得的声源增益来平滑噪声段中的声源增益。平滑量限制电路计算由声源增益除以已经平滑的声源增益和声源增益之间的差的绝对值所表示的波动量，并且限制平滑增益的值以使得波动量不会超过某一阈值的方式。

5. 发明授权

US08374852B2 Apparatus and method of code conversion and recording medium that records program for computer to execute the method 失效
标题翻译：代码转换和记录介质的设备和方法，用于记录计算机执行方法的程序
公开(公告)号：US08374852B2
公开(公告)日：2013-02-12
申请号：US11376436
申请日：2006-03-16
申请人： Atsushi Murashima
发明人： Atsushi Murashima
IPC分类号： G10L11/06 , G10L19/12 , G10L11/00 , G10L19/00 , G10L21/04
CPC分类号： G10L19/173 , G10L25/78
摘要： Disclosed is a code conversion method to convert a first code sequence conforming to a first speech coding scheme into a second code sequence conforming to a second speech coding scheme. The method includes the following steps. The first step discriminates whether the first code sequence corresponds to a speech part or to a non-speech part, and generates a numerical value that indicates the discrimination result as a control flag. The second step converts the first code sequence into the second code sequence and outputs said second code sequence, when the value of the control flag corresponds to the speech part. The third step outputs the second code sequence that corresponds to the value of the control flag, when the value of the control flag corresponds to the non-speech part.
摘要翻译：公开了一种将符合第一语音编码方案的第一代码序列转换为符合第二语音编码方案的第二代码序列的代码转换方法。该方法包括以下步骤。第一步骤鉴别第一代码序列是否对应于语音部分或非语音部分，并且产生指示鉴别结果作为控制标志的数值。当控制标志的值对应于语音部分时，第二步骤将第一代码序列转换为第二代码序列并输出所述第二代码序列。当控制标志的值对应于非语音部分时，第三步骤输出与控制标志的值对应的第二代码序列。

6. 发明授权

US07426465B2 Speech signal decoding method and apparatus using decoded information smoothed to produce reconstructed speech signal to enhanced quality 有权
标题翻译：语音信号解码方法和使用被解码的信息进行平滑以产生重构语音信号以提高质量的装置
公开(公告)号：US07426465B2
公开(公告)日：2008-09-16
申请号：US11335739
申请日：2006-01-20
申请人： Atsushi Murashima
发明人： Atsushi Murashima
IPC分类号： G10L21/02
CPC分类号： G10L19/083
摘要： In a speech signal decoding method, information containing at least a sound source signal, gain, and filter coefficients is decoded from a received bit stream. Voiced speech and unvoiced speech of a speech signal are identified using the decoded information. Smoothing processing based on the decoded information is performed for at least either one of the decoded gain and decoded filter coefficients in the unvoiced speech. The speech signal is decoded by driving a filter having the decoded filter coefficients by an excitation signal obtained by multiplying the decoded sound source signal by the decoded gain using the result of the smoothing processing. A speech signal decoding apparatus is also disclosed.
摘要翻译：在语音信号解码方法中，至少包含声源信号，增益和滤波系数的信息从接收到的比特流解码。使用解码的信息来识别语音信号的语音和无声语音。针对无声语音中的解码增益和解码滤波器系数中的至少一个，执行基于解码信息的平滑化处理。通过使用通过使用平滑处理的结果将解码的声源信号乘以解码的增益获得的激励信号来驱动具有解码的滤波器系数的滤波器来解码语音信号。还公开了语音信号解码装置。

7. 发明授权

US07050968B1 Speech signal decoding method and apparatus using decoded information smoothed to produce reconstructed speech signal of enhanced quality 有权
标题翻译：语音信号解码方法和使用解码信息进行平滑以产生增强质量的重构语音信号的装置
公开(公告)号：US07050968B1
公开(公告)日：2006-05-23
申请号：US09627421
申请日：2000-07-27
申请人： Atsushi Murashima
发明人： Atsushi Murashima
IPC分类号： G10L21/02
CPC分类号： G10L19/083
摘要： In a speech signal decoding method, information containing at least a sound source signal, gain, and filter coefficients is decoded from a received bit stream. Voiced speech and unvoiced speech of a speech signal are identified using the decoded information. Smoothing processing based on the decoded information is performed for at least either one of the decoded gain and decoded filter coefficients in the unvoiced speech. The speech signal is decoded by driving a filter having the decoded filter coefficients by an excitation signal obtained by multiplying the decoded sound source signal by the decoded gain using the result of the smoothing processing. A speech signal decoding apparatus is also disclosed.
摘要翻译：在语音信号解码方法中，至少包含声源信号，增益和滤波系数的信息从接收到的比特流解码。使用解码的信息来识别语音信号的语音和无声语音。针对无声语音中的解码增益和解码滤波器系数中的至少一个，执行基于解码信息的平滑化处理。通过使用通过使用平滑处理的结果将解码的声源信号乘以解码的增益获得的激励信号来驱动具有解码的滤波器系数的滤波器来解码语音信号。还公开了语音信号解码装置。

8. 发明授权

US07031913B1 Method and apparatus for decoding speech signal 有权
标题翻译：用于解码语音信号的方法和装置
公开(公告)号：US07031913B1
公开(公告)日：2006-04-18
申请号：US09658045
申请日：2000-09-08
申请人： Atsushi Murashima
发明人： Atsushi Murashima
IPC分类号： G10L19/04 , G10L19/14
CPC分类号： G10L19/083 , G10L21/0364 , G10L2019/0012
摘要： A method of decoding a speech signal based on a CELP (Code Excited Linear Prediction) with improvement in degradation of decoded sound quality in a noise period. The method includes the steps of: calculating a norm of an excitation vector for each fixed period in a noise period; smoothing the calculated norm using a norm obtained in a previous period; changing the amplitude of the excitation vector in the period using the calculated norm and the smoothed norm; and driving a synthesizing filter by the excitation vector with the changed amplitude.
摘要翻译：一种基于CELP（代码激励线性预测）对语音信号进行解码的方法，其改善了噪声周期中解码音质的劣化。该方法包括以下步骤：在噪声周期中计算每个固定周期的激励矢量的范数; 使用前一期间获得的标准平滑计算出的标准; 使用计算的范数和平滑的范数改变周期中的激励矢量的幅度; 并通过振幅变化的激励矢量驱动合成滤波器。

9. 发明申请

US20050131682A1 Method and apparatus for transcoding between different speech encoding/decoding systems and recording medium 失效
标题翻译：用于在不同语音编码/解码系统和记录介质之间进行代码转换的方法和装置
公开(公告)号：US20050131682A1
公开(公告)日：2005-06-16
申请号：US11039969
申请日：2005-01-24
申请人： Atsushi Murashima
发明人： Atsushi Murashima
IPC分类号： G10L19/08 , G10L19/12 , G10L19/14 , G10L19/10
CPC分类号： G10L19/083 , G10L19/12 , G10L19/173
摘要： Disclosed is a code converting apparatus for converting a first code sequence conforming to a first system to a second code sequence conforming to a second system, in which a speech decoding circuit acquires a first linear prediction coefficient and the information on an excitation signal from the first code sequence, and actuates a filter having the aforementioned first linear prediction coefficient with the excitation signal obtained from the information on the excitation signal, to generate a first speech signal. A gain code generating circuit calculates a gain minimizing the distance between a second speech signal, generated from the information, obtained from the second code sequence, and the first speech signal (optimum gain), and corrects the optimum gain and the gain code generating circuit then finds the gain information in the second code sequence, based on the optimum gain as corrected (optimum gain corrected), the above optimum gain and a gain read out from a gain codebook of the second system. The gain is found at this time, in a non-speech segment, based on a speech decision value, using an evaluation function which will reduce time variations of the gain of the second system.
摘要翻译：公开了一种代码转换装置，用于将符合第一系统的第一代码序列转换为符合第二系统的第二代码序列，其中语音解码电路获取第一线性预测系数和来自第一系统的激励信号的信息并且利用从关于激励信号的信息获得的激励信号来驱动具有上述第一线性预测系数的滤波器，以产生第一语音信号。增益代码产生电路计算使从第二代码序列获得的信息产生的第二语音信号与第一语音信号（最佳增益）之间的距离最小化的增益，并且校正最佳增益和增益代码生成电路然后基于经校正的最佳增益（最佳增益校正），上述最佳增益和从第二系统的增益码本读出的增益，找到第二代码序列中的增益信息。此时，在非语音段中，基于语音决定值，使用将减少第二系统的增益的时间变化的评估函数来获得增益。

10. 发明授权

US06865534B1 Speech and music signal coder/decoder 失效
标题翻译：语音和音乐信号编码器/解码器
公开(公告)号：US06865534B1
公开(公告)日：2005-03-08
申请号：US09719826
申请日：1999-06-15
申请人： Atsushi Murashima , Kazunori Ozawa
发明人： Atsushi Murashima , Kazunori Ozawa
IPC分类号： G10L19/04 , G10L11/00 , G10L13/02 , G10L19/00 , G10L19/06 , G10L19/12 , H03M7/30
CPC分类号： G10L19/12 , G10L19/0204 , G10L19/0212
摘要： It is an object of the invention to excellently code a speech and music signal over all of bands in a speech and music signal code decoding apparatus having a band divided constitution. In order to achieve the object, a residue vector is generated by using an inverse filter (230 of FIG. 3) from a difference vector outputted from a first differencer (180 of FIG. 3). A band selecting circuit (250 of FIG. 3) generates n pieces of subvectors by using a component included in an arbitrary band in the residue vector subjected to orthogonal transformation. An orthogonal transformation coefficient quantizing circuit (260 of FIG. 3) quantizes n pieces of the subvectors.
摘要翻译：本发明的目的是在具有带分割结构的语音和音乐信号编码解码装置中对所有频带上的语音和音乐信号进行卓越的编码。为了实现该目的，通过使用从第一差分器（图3的180）输出的差矢量使用逆滤波器（图3的230）来生成残差矢量。频带选择电路（图3的250）通过使用经过正交变换的残差矢量中的任意频带中包含的分量来生成n个子矢量。正交变换系数量化电路（图3的260）对n个子矢量进行量化。

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