公开(公告)号：US09288792B2

公开(公告)日：2016-03-15

申请号：US14123333

申请日：2012-05-25

申请人： Takako Hori ,  Hiroyuki Ehara

发明人： Takako Hori ,  Hiroyuki Ehara

IPC分类号： H04M1/00 ,  H04W72/04 ,  H04W76/04 ,  H04W88/18

CPC分类号： H04W36/0022 ,  H04W72/04 ,  H04W76/22 ,  H04W88/181

摘要： The objective of the present invention is to suppress deterioration of call quality caused by transcoding without interrupting a call even if a codec used by one of the terminals during communication is changed. A modification determination unit, in the case of detecting a modification of a codec used by one terminal of two terminals, determines whether or not to constrain the bandwidth of the first codec using a first codec of the other terminal and a second codec after modification by the first-mentioned one of the terminals. A signaling generation unit transmits, to the other terminal, signaling for limiting the bandwidth if the bandwidth is to be limited.

摘要翻译： 本发明的目的在于，即使通信中的终端之一使用的编解码器被改变，也可以抑制代码转换引起的呼叫质量的恶化，而不会中断呼叫。 在检测到由两个终端的一个终端使用的编解码器的修改的情况下，修改确定单元确定是否使用另一终端的第一编解码器限制第一编解码器的带宽，并且修改后由第二编解码器 首先提到的一个终端。 如果要限制带宽，则信令生成单元向另一终端发送用于限制带宽的信令。

公开(公告)号：US09264094B2

公开(公告)日：2016-02-16

申请号：US14123841

申请日：2012-05-25

申请人： Katsunori Daimou ,  Masahiro Oshikiri ,  Hiroyuki Ehara

发明人： Katsunori Daimou ,  Masahiro Oshikiri ,  Hiroyuki Ehara

IPC分类号： G10L19/00 ,  H04B1/66 ,  G10L19/24 ,  G10L19/02 ,  G10L21/038

CPC分类号： H04B1/667 ,  G10L19/02 ,  G10L19/0204 ,  G10L19/24 ,  G10L21/038

摘要： A voice coding device capable of preventing overall quality degradation even when the bit rate for coding is lowered. The voice coding device codes a wide band signal in a first layer, and codes an extended band signal whose frequency band is located in higher frequency than the wide band signal in an extended band layer. An adaptive band selection unit (301) selects a frequency band to be excluded from a coding object in the extended band layer or a frequency band whose energy is to be attenuated in the extended band layer. A band-limited signal generation unit (302) excludes, within the frequency band of an input signal, the frequency band selected by the adaptive band selection unit (301) from the coding object, or attenuates the energy of the frequency band selected by the adaptive band selection unit (301).

摘要翻译： 即使当编码的比特率降低时，能够防止整体质量劣化的语音编码装置。 语音编码装置对第一层中的宽带信号进行编码，并且对扩展频带信号进行编码，其频带位于比扩频带中的宽带信号更高的频带。 自适应频带选择单元（301）从扩展频带层中的编码对象或能量在扩展频带层中衰减的频带中选择要排除的频带。 频带限制信号生成单元（302）在输入信号的频带内从编码对象排除由自适应频带选择部（301）选择的频带，或者衰减由所述编码对象选择的频带的能量 自适应频带选择单元（301）。

公开(公告)号：US09129590B2

公开(公告)日：2015-09-08

申请号：US12528671

申请日：2008-02-29

申请人： Takuya Kawashima ,  Hiroyuki Ehara ,  Koji Yoshida

发明人： Takuya Kawashima ,  Hiroyuki Ehara ,  Koji Yoshida

IPC分类号： G10L19/00 ,  G10L19/005 ,  G10L19/12

CPC分类号： G10L19/005 ,  G10L19/12

摘要： Disclosed are an audio encoding device and an audio decoding device which reduce degradation of subjective quality of a decoding signal caused by power mismatch of a decoding signal which is generated by a concealing process upon disappearance of a frame. When a frame is lost, a past encoding parameter is used to obtain a concealed LPC of the current frame and a concealed sound source parameter. A normal CELP decoding is performed from the obtained concealed sound source parameter. Correction is performed by using a conceal parameter on the obtained concealed LPC and the concealed sound source signal. The power of the corrected concealed sound source signal is adjusted to match a reference sound source power. A filter gain of the synthesis filter is adjusted so as to adjust the power of a decoded sound signal to the power of a decoded sound signal during an error-free state. Moreover, a synthesis filter gain adjusting coefficient is calculated by using an estimated normalized residual power so that a filter gain of a synthesis filter formed by using a concealed LPC is a filter gain during an error-free state.

摘要翻译： 公开了一种音频编码装置和音频解码装置，其减少由于帧消失时由隐藏处理产生的解码信号的功率失配引起的解码信号的主观质量的劣化。 当帧丢失时，使用过去的编码参数来获得当前帧的隐藏的LPC和隐藏的声源参数。 从获得的隐藏声源参数执行正常的CELP解码。 通过对所获得的隐蔽LPC和隐藏的声源信号使用隐藏参数来执行校正。 校正的隐蔽声源信号的功率被调整以匹配参考声源功率。 调整合成滤波器的滤波器增益，以便在无错误状态期间将解码声音信号的功率调整为解码声音信号的功率。 此外，通过使用估计的归一化剩余功率来计算合成滤波器增益调整系数，使得通过使用隐藏的LPC形成的合成滤波器的滤波器增益在无错误状态期间是滤波器增益。

公开(公告)号：US08812306B2

公开(公告)日：2014-08-19

申请号：US12307974

申请日：2007-07-11

申请人： Takuya Kawashima ,  Hiroyuki Ehara ,  Koji Yoshida

发明人： Takuya Kawashima ,  Hiroyuki Ehara ,  Koji Yoshida

IPC分类号： G10L21/00 ,  H04L1/00 ,  G10L19/005 ,  G10L25/90 ,  H04L12/56

CPC分类号： G10L19/005 ,  G10L25/90 ,  G10L2019/0011 ,  H04L47/10

摘要： An audio decoding device capable of suppressing an information amount for a lost flame compensation process and encoding efficiency is provided. A decoded sound source generator generates a lost frame's CELP decoded sound source signal. A pitch pulse information decoder CELP decodes a pitch pulse position information and a pitch pulse amplitude information. A pitch pulse waveform learner learns a pitch pulse learning waveform in a past frame in advance from the lost frame. A convolution adjuster amplitude-adjusts the pitch pulse learning waveform according to the pitch pulse amplitude information by considering a predetermined number of waveforms peripheral to a peak position of the lost frame's CELP decoded excitation signal, and convolutes a pitch pulse waveform into a time axis which has been amplitude-adjusted according to the pitch pulse position information. A sound source signal corrector adds or replaces the pitch pulse waveform convoluted into the time axis to the lost flame decoded sound source signal.

摘要翻译： 提供能够抑制失火补偿处理和编码效率的信息量的音频解码装置。 解码声源发生器产生丢失帧的CELP解码声源信号。 音调脉冲信息解码器CELP解码音调脉冲位置信息和音调脉冲幅度信息。 音调脉冲波形学习者从丢失的帧中预先学习过去帧中的音调脉冲学习波形。 卷积调整器通过考虑丢失帧的CELP解码的激励信号的峰值位置外围的预定数量的波形，根据音调脉冲幅度信息来振幅调整音调脉冲学习波形，并将音调脉冲波形卷曲成时间轴， 已经根据音调脉冲位置信息进行了幅度调整。 声源信号校正器将与时间轴卷积的音调脉冲波形相加或替换为失火解码声源信号。

公开(公告)号：US08644526B2

公开(公告)日：2014-02-04

申请号：US12992791

申请日：2009-06-26

申请人： Hiroyuki Ehara ,  Takuya Kawashima ,  Koji Yoshida

发明人： Hiroyuki Ehara ,  Takuya Kawashima ,  Koji Yoshida

IPC分类号： H03G3/00

CPC分类号： G10L19/008

摘要： Fluctuation in decoded signal localization is suppressed to maintain the feel of stereo. A selection unit selects balance parameters when the balance parameters are input from a gain coefficient decoding unit, or selects balance parameters input from a gain coefficient calculation unit when there is no balance parameter input from the gain coefficient decoding unit, and outputs the selected balance parameters to a multiplication unit. The multiplication unit multiplies a gain coefficient input from the selection unit with a decoded monaural signal input from a monaural decoding unit to perform balance adjustment processing.

摘要翻译： 抑制解码信号定位的波动以保持立体感。 当从增益系数解码单元输入平衡参数时，选择单元选择平衡参数，或者当没有从增益系数解码单元输入的平衡参数时选择从增益系数计算单元输入的平衡参数，并输出所选择的平衡参数 到乘法单元。 乘法单元将从选择单元输入的增益系数与从单声道解码单元输入的解码单声道信号相乘，以进行平衡调整处理。

公开(公告)号：US08473288B2

公开(公告)日：2013-06-25

申请号：US12990697

申请日：2009-06-18

申请人： Toshiyuki Morii ,  Hiroyuki Ehara ,  Koji Yoshida

发明人： Toshiyuki Morii ,  Hiroyuki Ehara ,  Koji Yoshida

IPC分类号： G10L19/00

CPC分类号： G10L19/008 ,  G10L25/27

摘要： Disclosed are a quantizer, encoder, and the methods thereof, wherein the computational load is reduced when the values related to the transform coefficients of the principal component analysis transform are quantized when a principal component analysis transform is applied to code stereo. A quantizer includes a power correlation calculator which calculates the power of the left channel signal, the power of the right channel signal, and the correlation between the left channel signal and the right channel signal; an intermediate value calculator which calculates the intermediate value which is the difference between left channel signal the power and the right channel signal power; a codebook which holds a plurality of sets of the coefficients related to the transform coefficients of the principal component analysis transform and the code; and a quantizer which calculates the sum of the first multiplication result obtained by multiplying the coefficient by the correlation value and the second multiplication result obtained by multiplying the coefficient by the intermediate value as the cost function E, selects the coefficients where the cost function E becomes the maximum, and fetches the code related to the selected coefficients as the quantized code.

摘要翻译： 公开了一种量化器，编码器及其方法，其中当将主分量分析变换应用于代码立体声时，当与主成分分析变换的变换系数相关的值被量化时，计算负荷减小。 量化器包括功率相关计算器，其计算左声道信号的功率，右声道信号的功率，以及左声道信号和右声道信号之间的相关性; 中间值计算器，其计算作为左声道信号的功率和右声道信号功率之间的差的中间值; 码本，其保存与主成分分析变换和代码的变换系数相关的多个系数集合; 以及量化器，其计算通过将系数乘以相关值而获得的第一相乘结果与通过将系数乘以中间值而获得的第二乘法结果作为成本函数E的和，选择成本函数E变为 最大值，并且将与所选择的系数相关的代码作为量化代码获取。

公开(公告)号：US08370138B2

公开(公告)日：2013-02-05

申请号：US12293302

申请日：2007-03-15

申请人： Takuya Kawashima ,  Hiroyuki Ehara ,  Koji Yoshida

发明人： Takuya Kawashima ,  Hiroyuki Ehara ,  Koji Yoshida

IPC分类号： G10L19/02

CPC分类号： G10L19/005 ,  G10L19/002 ,  G10L19/24

摘要： A scalable encoding device is capable of improving quality of a decoded signal without increasing an encoding amount and compensating data with a sufficient quality upon data loss. An extension layer bit distribution calculator calculates a bit distribution of a quality improving encoding data and compensation encoding data in the extension layer according to an audio mode of the input signal. An extension layer encoder generates quality improving encoding data according to the specified number of bits. A compensation information encoder extracts a part of core layer encoding data and makes it as compensation encoding data for the core layer. An extension layer encoded data generator multiplexes the extension layer bit distribution information, the compensation encoding data, and the quality improving encoding data so as to obtain extension layer encoding data.

摘要翻译： 可扩展编码装置能够在不增加编码量的情况下提高解码信号的质量，并且在数据丢失时能够以足够的质量补偿数据。 扩展层位分配计算器根据输入信号的音频模式来计算扩展层中的质量改进编码数据和补偿编码数据的位分布。 扩展层编码器根据指定的位数生成质量改进编码数据。 补偿信息编码器提取核心层编码数据的一部分，并将其作为核心层的补偿编码数据。 扩展层编码数据生成器多路复用扩展层位分配信息，补偿编码数据和质量改进编码数据，以便获得扩展层编码数据。

公开(公告)号：US08239191B2

公开(公告)日：2012-08-07

申请号：US12440661

申请日：2007-09-14

申请人： Hiroyuki Ehara ,  Toshiyuki Morii ,  Koji Yoshida

发明人： Hiroyuki Ehara ,  Toshiyuki Morii ,  Koji Yoshida

IPC分类号： G10L19/00 ,  G10L21/02

CPC分类号： G10L19/265 ,  G10L19/08

摘要： Disclosed is an audio encoding device capable of adjusting a spectrum inclination of a quantized noise without changing the Formant weight. The device includes: an HPF (131) which extracts a high-frequency component of the frequency region from an input audio signal; a high-frequency energy level calculation unit (132) which calculates an energy level of the high-frequency component in a frame unit; an LPF (133) which extracts a low-frequency component of the frequency region from the input audio signal; a low-energy level calculation unit (134) which calculates an energy level of a low-frequency component in a frame unit; an inclination correction coefficient calculation unit (141) multiplies the difference between SNR of the high-frequency component and SNR of the low-frequency component inputted from an adder (140) by a constant and adds a bias component to the product so as to calculate an inclination correction coefficient ?3. The inclination correction coefficient is used for adjusting the spectrum inclination of a quantized noise.

摘要翻译： 公开了能够在不改变共振峰重量的情况下调整量化噪声的频谱倾斜度的音频编码装置。 该装置包括：HPF（131），其从输入音频信号中提取频率区域的高频分量; 高频能量计算单元，其计算帧单位中的高频分量的能级; LPF（133），其从所述输入音频信号中提取所述频率区域的低频分量; 低能量计算单元（134），其计算帧单位中的低频分量的能级; 倾斜校正系数计算单元（141）将从加法器（140）输入的低频分量的SNR与高频成分的SNR之差乘以常数，并将偏置分量加到乘积上，以计算 倾斜校正系数α3。 倾斜校正系数用于调整量化噪声的频谱倾斜度。

公开(公告)号：US08160868B2

公开(公告)日：2012-04-17

申请号：US11908513

申请日：2006-03-13

申请人： Takuya Kawashima ,  Hiroyuki Ehara

发明人： Takuya Kawashima ,  Hiroyuki Ehara

IPC分类号： G10L11/04

CPC分类号： G10L19/24 ,  G10L19/005 ,  H04L1/0057

摘要： A scalable decoder capable of avoiding deterioration in subjective quality of a listener. The scalable decoder for decoding core layer encoding data and extension layer encoding data including an extension layer gain coefficient, wherein a voice analysis section detects variation in power of a core layer decoding voice signal being obtained from the core layer encoding data, a gain attenuation rate calculating section (140) sets the attenuation intensity variable depending on variation in power, and a gain attenuation section (143) attenuates the extension layer gain coefficient in a second period preceding a first period according to a set attenuation intensity when extension layer encoding data in the first period is missing, thus interpolating the extension layer gain coefficient in the first period.

摘要翻译： 一种能够避免听众主观质量恶化的可扩展解码器。 用于解码核心层编码数据的可扩展解码器和包括扩展层增益系数的扩展层编码数据，其中语音分析部分检测从核心层编码数据获得的核心层解码语音信号的功率变化，增益衰减率 计算部分（140）根据功率变化设置衰减强度变化，并且增益衰减部分（143）根据设置的衰减强度在第一周期之前的第二周期中衰减扩展层增益系数，当扩展层编码数据 第一个周期丢失，因此在第一个周期内插入扩展层增益系数。

公开(公告)号：US08150684B2

公开(公告)日：2012-04-03

申请号：US11994140

申请日：2006-06-27

申请人： Takuya Kawashima ,  Hiroyuki Ehara

发明人： Takuya Kawashima ,  Hiroyuki Ehara

IPC分类号： G10L19/00

CPC分类号： G10L19/005 ,  G10L19/24

摘要： A scalable decoder capable of preventing degradation of the quality of the decoded signal in a disappeared data interpolation in band scalable coding. A core layer decoding section (101) acquires a core layer decoded signal and narrow band spectrum information by decoding. A narrow band spectrum slope computing section (103) computes the slope of an attenuation line of a narrow band spectrum from the narrow band spectrum information. An extended layer disappearance detection section (104) detects whether extended layer coded data has disappeared or not. An extended layer decoding section (105) normally decodes the extended layer coded data. If the extended layer disappears, a parameter required for decoding is interpolated and synthesizes an interpolation decoded signal by the interpolated parameter. The gain of the interpolated data is controlled according to the results of the computation, by the narrow band spectrum slope computing section (103).

摘要翻译： 一种可扩展的解码器，其能够防止频带可缩放编码中消失的数据插值中的解码信号的质量劣化。 核心层解码部（101）通过解码获取核心层解码信号和窄带频谱信息。 窄带频谱斜率计算部分（103）从窄带频谱信息计算窄带频谱的衰减线的斜率。 扩展层消失检测部（104）检测扩展层编码数据是否已经消失。 扩展层解码部（105）通常对扩展层编码数据进行解码。 如果扩展层消失，则内插解码所需的参数，并通过内插参数合成内插解码信号。 通过窄带频谱斜率计算部（103）根据计算结果来控制内插数据的增益。