会员体验
专利管家(专利管理)
工作空间(专利管理)
风险监控(情报监控)
数据分析(专利分析)
侵权分析(诉讼无效)
联系我们
交流群
官方交流:
QQ群: 891211   
微信请扫码    >>>
现在联系顾问~
热词
    • 71. 发明授权
    • Sound encoder and sound encoding method for generating a second layer decoded signal based on a degree of variation in a first layer decoded signal
    • 声音编码器和声音编码方法,用于基于第一层解码信号的变化程度来产生第二层解码信号
    • US08099275B2
    • 2012-01-17
    • US11577424
    • 2005-10-25
    • Masahiro Oshikiri
    • Masahiro Oshikiri
    • G06F15/00G10L11/00G10L19/00G10L11/04
    • G10L19/02
    • A sound encoder having an improved quantization performance while suppressing an increase of the bit rate to a lowest level. In a second layer encoder, a standard deviation calculator calculates a standard deviation σc of a first layer decoding spectrum after decoding a scale factor ratio multiplication and outputs the standard deviation σc to a selector. The selector selects a linear transform function as a function for a nonlinear transform of a residual spectrum according to the standard deviation σc A nonlinear transform function selects one of prepared nonlinear transform functions #1 to #N according to a result of the selection by the selector, and outputs the selected one to an inverse transformer. The inverse transformer subjects an inverse transform (expansion) to a residual spectrum candidate that is stored in a residual spectrum code book using the nonlinear transform function outputted from the nonlinear transform function and outputs the result to an adder.
    • 一种声音编码器,其具有改善的量化性能,同时抑制比特率增加到最低水平。 在第二层编码器中,标准偏差计算器在解码比例因子比乘法后计算第一层解码频谱的标准偏差σ,并将标准偏差&sgr; c输出到选择器。 选择器根据标准偏差选择线性变换函数作为残差谱的非线性变换的函数; c非线性变换函数根据选择结果选择准备的非线性变换函数#1至#N中的一个 选择器,并将所选择的一个输出到逆变换器。 逆变换器使用从非线性变换函数输出的非线性变换函数对存储在残差频谱代码簿中的残差频谱候选进行逆变换(扩展),并将结果输出到加法器。
    • 74. 发明申请
    • AUDIO DECODING DEVICE AND POWER ADJUSTING METHOD
    • 音频解码设备和功率调节方法
    • US20100332223A1
    • 2010-12-30
    • US12517603
    • 2007-12-12
    • Toshiyuki MoriiMasahiro Oshikiri
    • Toshiyuki MoriiMasahiro Oshikiri
    • G10L19/14
    • G10L19/26H03G3/3005
    • Provided is an audio decoding device capable of obtaining a preferable synthesized sound with a stable sound volume. The audio decoding device includes: a post filter (210) which performs a process for improving subjective quality of audio and a process for improving subjective quality of a steady-state noise on an output signal of a synthesis filter (209); an amplitude ratio/change amount calculation unit (211) which calculates the amplitude ratio of the input signal and the output signal of the post filter (210) and calculates the fluctuation amount of the amplitude ratio for each of sub-frames; a smoothing coefficient setting unit (212) sets a smoothing coefficient on each of the sub-frames by using the amplitude ratio of the input signal and the output signal of the post filter (210) and the fluctuation amount of the amplitude ratio; an adjustment coefficient setting unit (213) which sets an adjustment coefficient for each sample by using the amplitude ratio of the input signal and the output signal of the post filter (210) and the smoothing coefficient; and a power adjusting unit (214) which multiplies the output signal of the post filter (210) by the adjustment coefficient so as to adjust the power of the output signal of the post filter (210).
    • 提供能够获得具有稳定音量的优选合成声音的音频解码装置。 音频解码装置包括:后处理器(210),其执行用于提高音频的主观质量的处理和用于提高合成滤波器(209)的输出信号上的稳态噪声的主观质量的处理; 计算后置滤波器(210)的输入信号和输出信号的振幅比的振幅比/变化量计算单元(211),并计算每个子帧的振幅比的波动量; 平滑系数设定单元(212)通过使用后置滤波器(210)的输入信号和输出信号的振幅比和振幅比的变动量,对每个子帧设置平滑系数; 调整系数设定单元,通过使用后置滤波器(210)的输入信号和输出信号的幅度比和平滑系数来设定每个采样的调整系数; 以及功率调整单元(214),其将后置滤波器(210)的输出信号乘以调节系数,以便调整后置滤波器(210)的输出信号的功率。
    • 75. 发明授权
    • Encoder, decoder, encoding method, and decoding method
    • 编码器,解码器,编码方法和解码方法
    • US07769584B2
    • 2010-08-03
    • US11718452
    • 2005-11-02
    • Masahiro OshikiriHiroyuki EharaKoji Yoshida
    • Masahiro OshikiriHiroyuki EharaKoji Yoshida
    • G10L19/02
    • G10L21/038
    • An encoder, decoder, encoding method, and decoding method enabling acquisition of high-quality decoded signal in scalable encoding of an original signal in first and second layers even if the second or upper layer section performs low bit-rate encoding. In the encoder, a spectrum residue shape codebook (305) stores candidates of spectrum residue shape vectors, a spectrum residue gain codebook (307) stores candidates of spectrum residue gains, and a spectrum residue shape vector and a spectrum residue gain are sequentially outputted from the candidates according to the instruction from a search section (306). A multiplier (308) multiplies a candidate of the spectrum residue shape vector by a candidate of the spectrum residue gain and outputs the result to a filtering section (303). The filtering section (303) performs filtering by using a pitch filter internal state set by a filter state setting section (302), a lag T outputted by a lag setting section (304), and a spectrum residue shape vector which has undergone gain adjustment.
    • 即使第二或上层部分执行低比特率编码,编码器,解码器,编码方法和解码方法能够在第一和第二层中的原始信号的可分级编码中获取高质量解码信号。 在编码器中,频谱残差形状码本(305)存储频谱残差形状矢量的候补,频谱残差增益码本(307)存储频谱残差增益的候选,频谱残差形状矢量和频谱残差增益从 候选人根据来自搜索部分的指示(306)。 乘法器(308)将频谱残差形状矢量的候选乘以频谱残差增益的候选,并将结果输出到滤波部(303)。 滤波部(303)通过使用由滤波器状态设定部(302)设定的音调滤波器内部状态,由滞后设定部(304)输出的滞后T和进行了增益调整的频谱残差图形矢量进行滤波 。
    • 76. 发明申请
    • TRANSFORM CODER AND TRANSFORM CODING METHOD
    • 变换编码器和变换编码方法
    • US20090281811A1
    • 2009-11-12
    • US12089985
    • 2006-10-13
    • Masahiro OshikiriTomofumi Yamanashi
    • Masahiro OshikiriTomofumi Yamanashi
    • G10L19/00
    • G10L19/0208G10L19/038G10L19/24
    • A transform coder leading to reduction of degradation of perceptual sound quality even if an adequate number of bits is not assigned. Candidates of a correction scale factor stored in a correction scale factor codebook (123) are outputted one by one, and an error signal is generated by subjecting the candidate and scale factors outputted from scale factor computing sections (121, 122) to a predetermined operation. A judging section (126) determines a weight vector given to a weighted error computing section (127) depending on the sign of the error signal. The weighted error computing section (127) computes the square of the error signal, multiplies the square of the error signal by the weight vector given from the judging section (126), and computes a weighted squared error E. A search section (128) determines the candidates of the correction scale factor which minimizes the weighted squared error E by a closed loop processing.
    • 即使没有分配足够数量的比特,导致感知声音质量降低的变换编码器也是如此。 存储在校正比例因子码本(123)中的校正比例因子的候选者一个接一个地输出,并且通过将从比例因子计算部(121,122)输出的候选和比例因子进行预定的操作来生成错误信号 。 判断部(126)根据误差信号的符号确定给予加权误差运算部(127)的加权矢量。 加权误差计算部分(127)计算误差信号的平方,将误差信号的平方乘以从判断部分(126)给出的权重向量,并计算加权平方误差E.搜索部分(128) 通过闭环处理确定最小化加权平方误差E的校正比例因子的候选。
    • 77. 发明申请
    • CODING DEVICE AND CODING METHOD
    • 编码设备和编码方法
    • US20090094024A1
    • 2009-04-09
    • US12282287
    • 2007-03-08
    • Tomofumi YamanashiKaoru SatoToshiyuki MoriiMasahiro Oshikiri
    • Tomofumi YamanashiKaoru SatoToshiyuki MoriiMasahiro Oshikiri
    • G10L19/04G10L19/00
    • G10L19/24
    • A coding device is provided with features in which optimum coding in a higher layer is flexibly carried out based on a coding result of a lower layer and a quality audio signal in limited circumstances is served to users. In this coding device, a basic layer coding unit codes an input signal to generate a basic layer information source code and outputs a linear prediction coefficient (LPC) and a quantum LPC, which are parameters calculated at coding, to an expanded layer control unit. A basic layer decoding unit decodes the basic layer information source code. An adding unit reverses a polarity of a basic layer decoded signal, adds the same to the input signal, and calculates a difference signal. The expanded layer control unit generates expanded layer mode information indicative of a coding mode in an expanded layer based on the LPC and the quantum LPC. An expanded layer coding unit codes the difference signal obtained from the adding unit under control of the expanded layer control unit.
    • 编码装置具有这样的特征,其中基于较低层的编码结果灵活地执行较高层中的最佳编码,并且在有限的情况下向用户提供质量音频信号。 在该编码装置中,基本层编码单元对输入信号进行编码以生成基本层信息源代码,并将作为编码计算出的参数的线性预测系数(LPC)和量子LPC输出到扩展层控制单元。 基本层解码单元解码基本层信息源代码。 加法单元反转基本层解码信号的极性,将其相加于输入信号,并计算差分信号。 扩展层控制单元基于LPC和量子LPC生成表示扩展层中的编码模式的扩展层模式信息。 扩展层编码单元在扩展层控制单元的控制下对从加法单元获得的差异信号进行编码。
    • 78. 发明申请
    • Encoding Device, Decoding Device, and Method Thereof
    • 编码设备,解码设备及其方法
    • US20080262835A1
    • 2008-10-23
    • US11596254
    • 2005-05-17
    • Masahiro Oshikiri
    • Masahiro Oshikiri
    • G10L19/14
    • G10L19/02G10L19/032G10L21/038H04B1/667
    • There is disclosed an encoding device capable of improving similarity between the high frequency band spectrum of the original signal and a new spectrum to be generated while realizing a low bit rate when encoding a wide-band signal spectrum. The encoding device has sub-band amplitude calculation units (122, 128) for calculating the amplitude of the respective sub-bands for the high frequency band spectrum obtained from the wide-band signal. A search unit (124) and a gain codebook (125) select some sub-bands from a plurality of sub-bands and only the gain of the selected sub-bands is subjected to encoding. An interpolation unit (126) expresses the gain of the sub-band not selected, by mutually interpolating the selected gains.
    • 公开了一种编码装置,其能够在对宽带信号频谱进行编码时,在实现低比特率的同时,提高原始信号的高频带和要产生的新频谱之间的相似度。 编码装置具有子带幅度计算单元(122,128),用于计算从宽带信号获得的用于高频带频谱的各个子带的幅度。 搜索单元(124)和增益码本(125)从多个子带中选择一些子带,并且仅对所选子带的增益进行编码。 内插单元(126)通过相互插入所选择的增益来表示未选择的子带的增益。
    • 79. 发明申请
    • Sound Encoding Device And Sound Encoding Method
    • 声音编码装置和声音编码方法
    • US20080065373A1
    • 2008-03-13
    • US11577638
    • 2005-10-25
    • Masahiro Oshikiri
    • Masahiro Oshikiri
    • G10L19/02
    • G10L19/0212G10L19/022
    • A sound encoding device enabling the amount of delay to be kept small and the distortion between frames to be mitigated. In the sound encoding device, a window multiplication part (211) of a long analysis section (21) multiplies a long analysis frame signal of analysis length M1 by an analysis window, the resultant signal multiplied by the analysis window is outputted to an MDCT section (212), and the MDCT section (212) performs MDCT of the input signal to obtain the transform coefficients of the long analysis frame and outputs it to a transform coefficient encoding section (30). The window multiplication part (221) of a short analysis section (22) multiplies a short analysis frame signal of analysis length M2 (M2
    • 能够使延迟量保持较小并且帧之间的失真得到缓解的声音编码装置。 在声音编码装置中,长分析部(21)的窗乘法部(211)将分析长度M 1的长分析帧信号乘以分析窗,将乘以分析窗的合成信号输出到MDCT 部分(212)和MDCT部分(212)执行输入信号的MDCT以获得长分析帧的变换系数,并将其输出到变换系数编码部分(30)。 短分析部(22)的窗乘法部(221)将分析长度M 2(M 2
    • 80. 发明申请
    • Encoder, Decoder, Encoding Method, and Decoding Method
    • 编码器,解码器,编码方法和解码方法
    • US20080052066A1
    • 2008-02-28
    • US11718452
    • 2005-11-02
    • Masahiro OshikiriHiroyuki EharaKoji Yoshida
    • Masahiro OshikiriHiroyuki EharaKoji Yoshida
    • G10L19/12
    • G10L21/038
    • An encoder, decoder, encoding method, and decoding method enabling acquisition of high-quality decoded signal in scalable encoding of an original signal in first and second layers even if the second or upper layer section performs low bit-rate encoding. In the encoder, a spectrum residue shape codebook (305) stores candidates of spectrum residue shape vectors, a spectrum residue gain codebook (307) stores candidates of spectrum residue gains, and a spectrum residue shape vector and a spectrum residue gain are sequentially outputted from the candidates according to the instruction from a search section (306). A multiplier (308) multiplies a candidate of the spectrum residue shape vector by a candidate of the spectrum residue gain and outputs the result to a filtering section (303). The filtering section (303) performs filtering by using a pitch filter internal state set by a filter state setting section (302), a lag T outputted by a lag setting section (304), and a spectrum residue shape vector which has undergone gain adjustment.
    • 即使第二或上层部分执行低比特率编码,编码器,解码器,编码方法和解码方法能够在第一和第二层中的原始信号的可分级编码中获取高质量解码信号。 在编码器中,频谱残差形状码本(305)存储频谱残差形状矢量的候补,频谱残差增益码本(307)存储频谱残差增益的候选,频谱残差形状矢量和频谱残差增益从 候选人根据来自搜索部分的指示(306)。 乘法器(308)将频谱残差形状矢量的候选乘以频谱残差增益的候选,并将结果输出到滤波部(303)。 滤波部(303)通过使用由滤波器状态设定部(302)设定的音调滤波器内部状态,由滞后设定部(304)输出的滞后T和进行了增益调整的频谱残差图形矢量进行滤波 。