Coding Apparatus and Decoding Apparatus
    71.
    发明申请
    Coding Apparatus and Decoding Apparatus 有权
    编码装置和解码装置

    公开(公告)号:US20100138219A1

    公开(公告)日:2010-06-03

    申请号:US12700583

    申请日:2010-02-04

    申请人: Masahiro OSHIKIRI

    发明人: Masahiro OSHIKIRI

    IPC分类号: G10L19/00

    CPC分类号: H04B1/667 G10L19/083

    摘要: A coding apparatus capable of coding a spectrum at a low bit rate and with high quality without producing any disturbance in a harmonic structure of the spectrum. In this apparatus, internal state setting section sets an internal state of a filtering section using a first spectrum S1(k). A pitch coefficient setting section outputs a pitch coefficient T by gradually changing it. The filtering section calculates an estimated value S′2(k) of a second spectrum S2(k) based on a pitch coefficient T. A search section calculates the degree of similarity between S2(k) and S′2(k). At this time, pitch coefficient T′ corresponding to the maximum calculated degree of similarity is given to a filter coefficient calculation section. The filter coefficient calculation section determines a filter coefficient βi using this pitch coefficient T′.

    摘要翻译: 一种能够以低比特率和高质量对频谱进行编码,而不会在频谱的谐波结构中产生任何干扰的编码装置。 在该装置中,内部状态设定部使用第一频谱S1(k)设定滤波部的内部状态。 俯仰系数设定部通过逐渐变化来输出俯仰系数T. 滤波部基于音调系数T计算第二频谱S2(k)的估计值S'2(k)。搜索部分计算S2(k)和S'2(k)之间的相似度。 此时,将与最大计算出的相似度对应的音调系数T'赋予滤波器系数计算部。 滤波器系数计算部使用该音调系数T'来确定滤波器系数bgr i。

    TRANSFORM CODER AND TRANSFORM CODING METHOD
    72.
    发明申请
    TRANSFORM CODER AND TRANSFORM CODING METHOD 有权
    变换编码器和变换编码方法

    公开(公告)号:US20090281811A1

    公开(公告)日:2009-11-12

    申请号:US12089985

    申请日:2006-10-13

    IPC分类号: G10L19/00

    摘要: 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.

    摘要翻译: 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。

    CODING DEVICE AND CODING METHOD
    73.
    发明申请
    CODING DEVICE AND CODING METHOD 有权
    编码设备和编码方法

    公开(公告)号:US20090094024A1

    公开(公告)日:2009-04-09

    申请号:US12282287

    申请日:2007-03-08

    IPC分类号: G10L19/04 G10L19/00

    CPC分类号: G10L19/24

    摘要: 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.

    摘要翻译: 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。

    Encoding Device, Decoding Device, and Method Thereof
    74.
    发明申请
    Encoding Device, Decoding Device, and Method Thereof 有权
    编码设备,解码设备及其方法

    公开(公告)号:US20080262835A1

    公开(公告)日:2008-10-23

    申请号:US11596254

    申请日:2005-05-17

    申请人: Masahiro Oshikiri

    发明人: Masahiro Oshikiri

    IPC分类号: G10L19/14

    摘要: 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.

    摘要翻译: 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。

    Sound Encoding Device And Sound Encoding Method
    75.
    发明申请
    Sound Encoding Device And Sound Encoding Method 有权
    声音编码装置和声音编码方法

    公开(公告)号:US20080065373A1

    公开(公告)日:2008-03-13

    申请号:US11577638

    申请日:2005-10-25

    申请人: Masahiro Oshikiri

    发明人: Masahiro Oshikiri

    IPC分类号: G10L19/02

    CPC分类号: G10L19/0212 G10L19/022

    摘要: 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

    摘要翻译: 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

    Encoder, Decoder, Encoding Method, and Decoding Method
    76.
    发明申请
    Encoder, Decoder, Encoding Method, and Decoding Method 有权
    编码器,解码器,编码方法和解码方法

    公开(公告)号:US20080052066A1

    公开(公告)日:2008-02-28

    申请号:US11718452

    申请日:2005-11-02

    IPC分类号: G10L19/12

    CPC分类号: G10L21/038

    摘要: 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.

    摘要翻译: 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 。

    Method and system for speech encoding involving analyzing search range for current period according to length of preceding pitch period
    77.
    发明授权
    Method and system for speech encoding involving analyzing search range for current period according to length of preceding pitch period 有权
    用于语音编码的方法和系统,涉及根据前一音调周期的长度分析当前周期的搜索范围

    公开(公告)号:US06470310B1

    公开(公告)日:2002-10-22

    申请号:US09407060

    申请日:1999-09-28

    IPC分类号: G10L1104

    CPC分类号: G10L25/90 G10L19/08

    摘要: Processing for producing encoded output representing information about a pitch period of an input speech signal is performed. The pitch period of a previously entered speech signal is stored in a buffer. A search range-determining portion determines a range in which a current pitch period is analyzed, according to the pitch period of the previously entered speech signal. A presently entered speech signal is applied from a speech input terminal. A pitch analysis portion makes a pitch analysis of candidates for the pitch period contained in the determined search range. Information about the pitch period is delivered from an output terminal and stored in the buffer for subsequent processing. The pitch period of the speech signal can be calculated with a small amount of calculation and represented with a small amount of information.

    摘要翻译: 执行用于产生表示关于输入语音信号的音调周期的信息的编码输出的处理。 先前输入的语音信号的音调周期被存储在缓冲器中。 搜索范围确定部分根据先前输入的语音信号的音调周期来确定当前音调周期被分析的范围。 从语音输入端子应用当前输入的语音信号。 音调分析部分对包含在确定的搜索范围内的音调周期的候选进行音调分析。 关于音调周期的信息从输出端传送并存储在缓冲器中用于后续处理。 可以用少量的计算来计算语音信号的音调周期,并用少量的信息表示。

    Speech/audio encoding apparatus, speech/audio decoding apparatus, and methods thereof
    78.
    发明授权
    Speech/audio encoding apparatus, speech/audio decoding apparatus, and methods thereof 有权
    语音/音频编码装置,语音/音频解码装置及其方法

    公开(公告)号:US09536534B2

    公开(公告)日:2017-01-03

    申请号:US14001977

    申请日:2012-03-19

    摘要: Provided is a speech/audio encoding apparatus with which it is possible to code a significant frequency domain region with high precision, and to enable high audio quality. A speech/audio encoding apparatus codes a linear prediction coefficient. A significant frequency domain region detection unit identifies a frequency domain region which is aurally significant from the linear prediction coefficient. A frequency domain region repositioning unit repositions the significant frequency domain region which is identified by the significant frequency domain region detection unit. A bit allocation computation unit determines a coding bit allocation on the basis of the significant frequency domain region which is repositioned by the frequency domain region repositioning unit.

    摘要翻译: 提供了一种语音/音频编码装置,其可以以高精度对重要的频域区域进行编码,并且能够实现高音频质量。 语音/音频编码装置对线性预测系数进行编码。 显着的频域区域检测单元从线性预测系数识别听觉显着的频域区域。 频域区域重新定位单元重新定位由有效频域区域检测单元识别的显着频域区域。 比特分配计算单元基于由频域区域重定位单元重定位的有效频域区域来确定编码比特分配。

    Encoding device, decoding device, and methods therefor
    79.
    发明授权
    Encoding device, decoding device, and methods therefor 有权
    编码装置,解码装置及其方法

    公开(公告)号:US09111527B2

    公开(公告)日:2015-08-18

    申请号:US13318552

    申请日:2010-05-19

    申请人: Masahiro Oshikiri

    发明人: Masahiro Oshikiri

    IPC分类号: G10L19/008

    CPC分类号: G10L19/008

    摘要: An encoding device, a decoding device, and encoding and decoding methods are provided, wherein when a multi-channel signal is encoded with high efficiency, using an adaptive filter, the number of arithmetic operations to update a filter coefficient of the adaptive filter can be reduced. An update range determination unit determines the range of a filter coefficient order (update order range) of a filter coefficient to be updated, among filter coefficients gk(n) of the adaptive filter, on the basis of a mutual correlation function between an input (L) signal and an input (R) signal. The adaptive filter updates the filter coefficient gk(n) of the filter coefficient order (n) to be updated, using a decoding (L) signal and a decoding error (R) signal.

    摘要翻译: 提供了一种编码装置,解码装置以及编码和解码方法,其中当使用自适应滤波器以高效率对多信道信号进行编码时,更新自适应滤波器的滤波器系数的算术运算的数量可以是 减少 更新范围确定单元基于自适应滤波器的滤波器系数gk(n)中的相互相关函数来确定要更新的滤波器系数的滤波器系数顺序(更新顺序范围)的范围, L)信号和输入(R)信号。 自适应滤波器使用解码(L)信号和解码误差(R)信号来更新要更新的滤波器系数序列(n)的滤波器系数g k(n)。

    Encoding device and encoding method
    80.
    发明授权
    Encoding device and encoding method 有权
    编码设备和编码方法

    公开(公告)号:US08760323B2

    公开(公告)日:2014-06-24

    申请号:US13822823

    申请日:2011-09-07

    IPC分类号: H03M13/00 H03M7/30

    摘要: An encoding device enables the amount of processing operations to be significantly reduced while minimizing deterioration in the quality of an output signal. This encoding device (101) encodes an input signal by determining the correlation between a first signal generated by using the input signal and a second signal generated by a predetermined method. An importance assessment unit (202) sets the importance of each of a plurality of processing units obtained by dividing the frames of the input signal. A CELP coder (203) performs sparse processing in which the amplitude value of a predetermined number of samples among multiple samples constituted by the first signal and/or the second signal in each processing unit is set to zero according to the importance that was set for each processing unit, and calculates the correlation between the first signal and the second signal, either of which was subjected to sparse processing.

    摘要翻译: 编码装置使得可以显着减少处理操作量,同时最小化输出信号质量的劣化。 该编码装置(101)通过确定通过使用输入信号产生的第一信号和通过预定方法产生的第二信号之间的相关性来对输入信号进行编码。 重要性评估单元(202)设置通过划分输入信号的帧而获得的多个处理单元中的每个处理单元的重要性。 CELP编码器(203)执行稀疏处理,其中根据为每个处理单元中的第一信号和/或第二信号构成的多个样本中的预定数量样本的振幅值被设置为零,根据为 每个处理单元,并且计算第一信号和第二信号之间的相关性,其中任一个都进行稀疏处理。