公开(公告)号：US20110064229A1

公开(公告)日：2011-03-17

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： H04R5/00

CPC分类号： G10L19/008

摘要： Fluctuation in decoded signal localization is suppressed to maintain the feel of stereo. A selection unit (220) selects balance parameters when the balance parameters are input from a gain coefficient decoding unit (210), or selects balance parameters input from a gain coefficient calculation unit (223) when there is no balance parameter input from the gain coefficient decoding unit (210), and outputs the selected balance parameters to a multiplication unit (221). The multiplication unit (221) multiplies a gain coefficient input from the selection unit (220) with a decoded monaural signal input from a monaural decoding unit (202) 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（210）输入平衡参数时，选择单元（220）选择平衡参数，或者当没有从增益系数输入的平衡参数时，选择从增益系数计算单元（223）输入的平衡参数 解码单元（210），并将所选择的天平参数输出到乘法单元（221）。 乘法单元（221）将从选择单元（220）输入的增益系数与从单声道解码单元（202）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：US20090326930A1

公开(公告)日：2009-12-31

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： G10L11/04

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

摘要： Provided is an audio decoding device capable of suppressing an information amount for a lost frame compensation process and encoding efficiency. In this device, a decoded sound source generation unit (203) generates a lost frame decoded sound source signal; a pitch pulse information decoding unit (204) decodes the pitch pulse position information and the pitch pulse amplitude information; a pitch pulse waveform learning unit (205) learns a pitch pulse learning waveform in the past frame in advance from the lost frame; a convolution unit (206) amplitude-adjusts the pitch pulse learning waveform according to the pitch pulse amplitude information, and convolutes the pitch pulse waveform into a time axis which has been amplitude-adjusted according to the pitch pulse position information; a sound source signal correction unit (207) adds or replaces the pitch pulse waveform convoluted into the time axis to the lost frame decoded sound source signal.

摘要翻译： 提供了能够抑制丢失帧补偿处理和编码效率的信息量的音频解码装置。 在该装置中，解码声源生成部（203）生成丢失帧解码声源信号， 音调脉冲信息解码单元（204）对音调脉冲位置信息和音调脉冲幅度信息进行解码; 音调脉冲波形学习单元（205）从丢失帧中预先学习过去帧中的音调脉冲学习波形; 卷积单元（206）根据音调脉冲幅度信息对音调脉冲学习波形进行幅度调整，并且将音调脉冲波形与根据音调脉冲位置信息进行了幅度调整的时间轴进行卷积; 声源信号校正单元（207）将对时间轴进行卷积的音调脉冲波形相加或替换为丢失帧解码声源信号。

公开(公告)号：US20100049509A1

公开(公告)日：2010-02-25

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 ,  G10L19/08

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为滤波器增益。

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：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：US20090070107A1

公开(公告)日：2009-03-12

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 ,  H04L27/00

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

摘要： Provided is a scalable encoding device capable of improving quality of a decoded signal without increasing an encoding amount and compensating data with a sufficient quality upon data loss. In the scalable encoding device, an extension layer bit distribution calculation unit (103) 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 encoding unit (105) generates quality improving encoding data according to the specified number of bits. A compensation information encoding unit (104) extracts a part of core layer encoding data and makes it as compensation encoding data for the core layer. An extension layer encoded data generation unit (106) 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： 提供了一种可扩展编码装置，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。 在可扩展编码装置中，扩展层位分配计算单元（103）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配。 扩展层编码单元（105）根据指定的比特数产生质量改进编码数据。 补偿信息编码单元（104）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单元（106）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，以获得扩展层编码数据。

公开(公告)号：US08160874B2

公开(公告)日：2012-04-17

申请号：US12159312

申请日：2006-12-26

申请人： Takuya Kawashima ,  Hiroyuki Ehara

发明人： Takuya Kawashima ,  Hiroyuki Ehara

IPC分类号： G10L19/12 ,  G10L19/00

CPC分类号： G10L19/265 ,  G10L19/005 ,  G10L21/02

摘要： An audio decoding device performs frame loss compensation capable of obtaining a decoded audio which is natural for ears with little noise. The audio decoding device includes a non-cyclic pulse waveform detection unit for detecting a non-cyclic pulse waveform section in a n−1-th frame, which is repeatedly used with a pitch cycle in the n-th frame upon compensation of loss of the n-th frame. The audio coding device also includes a non-cyclic pulse waveform suppression unit for suppressing a non-cyclic pulse waveform by replacing an audio source signal existing in the non-cyclic pulse waveform section in the n−1-th frame by a noise signal. The audio coding device further includes a synthesis filter for using a linear prediction coefficient decoded by an LPC decoding unit to perform synthesis by a synthesis filter by using the audio source signal of the n−1-th frame from the non-cyclic pulse waveform suppression unit as a drive audio source, thereby obtaining the decoded audio signal of the n-th frame.

摘要翻译： 音频解码装置执行能够获得对具有小噪声的耳朵自然的解码音频的帧丢失补偿。 音频解码装置包括：非周期性脉搏波形检测单元，用于检测第1帧中的非循环脉搏波形部分，在第n帧中的音调周期被重复使用， 第n帧。 音频编码装置还包括非循环脉冲波形抑制单元，用于通过用噪声信号替换存在于第n-1帧中的非循环脉搏波形段中的音频源信号来抑制非循环脉冲波形。 音频编码装置还包括合成滤波器，用于使用由LPC解码单元解码的线性预测系数，通过使用来自非循环脉冲波形抑制的第n-1帧的音频源信号通过合成滤波器进行合成 单元作为驱动音频源，从而获得第n帧的解码音频信号。

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。