公开(公告)号：US07848932B2

公开(公告)日：2010-12-07

申请号：US11719413

申请日：2005-11-28

Applicant: Michiyo Goto ,  Koji Yoshida

Inventor： Michiyo Goto ,  Koji Yoshida

IPC: G10L19/00

CPC classification number: G10L19/008

Abstract: A stereo audio encoding apparatus capable of preventing degradation of the sound quality of a decoded signal, while reducing the encoding bit rate. A spatial information analyzer analyzes spatial information for each of a L channel signal and an R channel signal. A similarity raiser corrects, based on an analysis result of the spatial information analyzer, a difference between the spatial information of the L channel signal and that of the R channel signal, to raise a similarity between the L and R channel signals. A channel signal encoder uses a sound source common to the two channels to encode the L and R channel signals as raised in similarity and output the resultant single encoded information. A spatial information encoder encodes the analysis result of the spatial information analyzer to output the resultant encoded information.

Abstract translation: 一种能够在降低编码比特率的同时防止解码信号的声音质量下降的立体声音频编码装置。 空间信息分析器分析L信道信号和R信道信号中的每一个的空间信息。 相似性提升器根据空间信息分析器的分析结果，校正L信道信号的空间信息与R信道信号的空间信息之间的差异，以提高L信道和R信道信号之间的相似度。 信道信号编码器使用两个信道共同的声源来编码相似度上升的L和R信道信号，并输出所得到的单个编码信息。 空间信息编码器对空间信息分析器的分析结果进行编码，以输出所得到的编码信息。

公开(公告)号：US20100262421A1

公开(公告)日：2010-10-14

申请号：US12740727

申请日：2008-11-04

Applicant: Kok Seng Chong ,  Koji Yoshida ,  Masahiro Oshikiri

Inventor： Kok Seng Chong ,  Koji Yoshida ,  Masahiro Oshikiri

IPC: G10L19/02 ,  G10L19/00 ,  G10L19/14

CPC classification number: G10L19/008 ,  G10L19/0208 ,  G10L19/0212 ,  G10L19/08 ,  G10L19/24

Abstract: Provided is an encoding device which improves the sound quality of a stereo signal while maintaining a low bit rate. The encoding device includes: an LP inverse filter (121) which LP-inverse-filterS a left signal L(n) by using an inverse quantization linear prediction coefficient AdM(z) of a monaural signal; a T/F conversion unit (122) which converts the left sound source signal Le(n) from a temporal region to a frequency region; an inverse quantizer (123) which inverse-quantizes encoded information Mqe; spectrum division units (124, 125) which divide a high-frequency component of the sound source signal Mde(f) and the left signal Le(f) into a plurality of bands; and scale factor calculation units (126, 127) which calculate scale factors ai and ssi by using a monaural sound source signal Mdeh,i(f), a left sound source signal Leh,i(f), Mdeh,i(f), and right sound source signal Reh,i(f) of each divided band.

Abstract translation: 提供了一种在保持低比特率的同时改善立体声信号的声音质量的编码装置。 编码装置包括：LP逆滤波器，其通过使用单声道信号的逆量化线性预测系数AdM（z）对左信号L（n）进行反转滤波; 将左声源信号Le（n）从时间区域转换为频率区域的T / F转换单元（122） 逆量化器（123），其对编码信息Mqe进行逆量化; 将声源信号Mde（f）和左信号Le（f）的高频分量分割为多个频带的频谱分割单元（124,125） 以及通过使用单声道声源信号Mdeh，i（f），左声源信号Leh，i（f），Mdeh，i（f），...，f（f）来计算比例因子ai和ssi的比例因子计算单元（126,127） 和每个分频带的右声源信号Reh，i（f）。

公开(公告)号：US07783480B2

公开(公告)日：2010-08-24

申请号：US11574783

申请日：2005-09-15

Applicant: Koji Yoshida

Inventor： Koji Yoshida

IPC: G10L19/12 ,  G10L19/00

CPC classification number: G10L19/24 ,  G10L19/12

Abstract: An audio encoding apparatus and the like are disclosed which can improve the sound quality of encoded audio signals even in a case of scalable CELP encoding the audio signals in sections that vary with time. In this apparatus, an enhancement layer extended adaptive codebook generating part (102) generates an extended adaptive codebook (d_enh_ext[i]) from both one frame of core layer drive sound source signals (exc_core[n]) received from a core layer CELP encoding part (101) and past enhancement layer drive sound source signals (exc_enh[n]) received from an adder (106), and further inputs the generated extended adaptive codebook (d_enh_ext[i]) to an enhancement layer extended adaptive codebook (103) for each of sub-frames. That is, the enhancement layer extended adaptive codebook generating part (102) updates the extended adaptive codebook (d_enh_ext[i]) for each of the sub-frames.

Abstract translation: 公开了音频编码装置等，即使在随时间变化的部分中的音频信号进行可缩放CELP编码的情况下，也可以提高编码音频信号的声音质量。 在该装置中，增强层扩展自适应码本生成部分（102）从从核心层CELP编码接收到的核心层驱动声源信号（exc_core [n]）的一帧生成扩展自适应码本（d_enh_ext [i]） 从加法器（106）接收的部分（101）和过去的增强层驱动声源信号（exc_enh [n]），并且还将所生成的扩展自适应码本（d_enh_ext [i]）输入到增强层扩展自适应码本（103） 对于每个子帧。 也就是说，增强层扩展自适应码本生成部分（102）针对每个子帧更新扩展自适应码本（d_enh_ext [i]）。

公开(公告)号：US07730417B2

公开(公告)日：2010-06-01

申请号：US11436580

申请日：2006-05-19

Applicant: Koji Yoshida

Inventor： Koji Yoshida

IPC: G06F3/00 ,  G06F3/14

CPC classification number: G06Q10/10 ,  G06F9/451 ,  H04L12/1813

Abstract: Upon reception of data updating information from a server apparatus (14) in a situation in which a presentation window is “closed” or “minimized” on a display unit (24) of each of terminal apparatuses (11-13), that presentation window is displayed on the display unit (24). The user who uses each of the terminal apparatuses (11-13) can always recognize a change in situation of the presentation system.

Abstract translation: 在终端设备（11-13）的显示单元（24）上呈现窗口被“关闭”或“最小化”的情况下，从服务器设备（14）接收到数据更新信息时，该呈现窗口 显示在显示单元（24）上。 使用每个终端装置（11-13）的用户总是能够识别呈现系统的情况的变化。

公开(公告)号：US20100090529A1

公开(公告)日：2010-04-15

申请号：US12525613

申请日：2008-02-25

Applicant: Koji Yoshida

Inventor： Koji Yoshida

IPC: H02M7/00

CPC classification number: H02J7/0068 ,  H02J7/1423 ,  H02J7/345 ,  H02J2001/008 ,  Y10T307/406

Abstract: An electric power source device includes a direct current voltage source 10, a capacitor 13 connected in series with direct current voltage source 10, a DC/DC converter 19 connected to supply and receive energy between direct current voltage source 10 and capacitor 13, and a high voltage load 15 connected to both ends of a series circuit of direct current voltage source 10 and capacitor 13. A control circuit 33 of DC/DC converter 19 supplies electric power from capacitor 13 to direct current voltage source 10 when electric power is supplied to high voltage load 15 so as to reduce an electric current to be output from direct current voltage source 10 and to lighten the burden imposed on the direct current voltage source.

Abstract translation: 电源装置包括直流电压源10，与直流电压源10串联连接的电容器13，连接到在直流电压源10和电容器13之间提供和接收能量的DC / DC转换器19，以及 连接到直流电压源10和电容器13的串联电路的两端的高压负载15.DC / DC转换器19的控制电路33将电力从电容器13提供给直流电压源10，当电力被供给到 高电压负载15，以便减少从直流电压源10输出的电流并且减轻施加在直流电压源上的负担。

公开(公告)号：US20100049509A1

公开(公告)日：2010-02-25

申请号：US12528671

申请日：2008-02-29

Applicant: Takuya Kawashima ,  Hiroyuki Ehara ,  Koji Yoshida

Inventor： Takuya Kawashima ,  Hiroyuki Ehara ,  Koji Yoshida

IPC: G10L19/00 ,  G10L19/08

CPC classification number: G10L19/005 ,  G10L19/12

Abstract: 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.

Abstract translation: 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：US20090319264A1

公开(公告)日：2009-12-24

申请号：US12373085

申请日：2007-07-11

Applicant: Koji Yoshida ,  Hiroyuki Ehara

Inventor： Koji Yoshida ,  Hiroyuki Ehara

IPC: G10L21/00

CPC classification number: G10L19/005

Abstract: Disclosed is a sound decoding device capable of improving the lost frame compensation performance and improving quality of the decoded sound. In this device, a rise frame sound source compensation unit (154) generates a compensation sound source signal when the current frame is a lost frame and a rise frame. An average sound source pattern update unit (156) updates the average sound source pattern held in an average sound source pattern holding unit (157) over a plurality of frames. When a frame is lost, an LPC synthesis unit (159) performs LPC synthesis on a decoded sound source signal by using the compensation sound source signal inputted via a switching unit (158) and a decoded LPC parameter from an LPC decoding unit (152) and outputs the compensation decoded sound signal.

Abstract translation: 公开了一种声音解码装置，其能够改善丢失的帧补偿性能并提高解码声音的质量。 在该装置中，上升帧声源补偿单元（154）当当前帧是丢帧和上升帧时产生补偿声源信号。 平均声源图案更新单元（156）通过多个帧来更新保持在平均声源图案保持单元（157）中的平均声源图案。 当帧丢失时，LPC合成单元（159）通过使用经由切换单元（158）输入的补偿声源信号和来自LPC解码单元（152）的解码LPC参数，对解码声源信号进行LPC合成， 并输出补偿解码声音信号。

公开(公告)号：US20090281795A1

公开(公告)日：2009-11-12

申请号：US12089814

申请日：2006-10-13

Applicant: Hiroyuki Ehara ,  Koji Yoshida

Inventor： Hiroyuki Ehara ,  Koji Yoshida

IPC: G10L21/00 ,  G10L19/12

CPC classification number: G10L19/24 ,  G10L19/26

Abstract: There is provided an audio encoding device for correcting the component having insufficient encoding capability in the core layer by an extended layer. In this device, a core layer encoding unit (101) encodes an audio signal, an extended layer encoding unit (150) encodes an encoding residual of the core layer encoding unit (101), a characteristic correction inverse filter (102 arranged at the pre-stage of an LPC synthesis filter (104) subjects the component having insufficient encoding capability in the core layer to the inverse characteristic correction process, and a characteristic correction filter (105) arranged at the post-stage of the LPC synthesis filter (104) performs a process for characteristic correction of the synthesis signal inputted from the LPC synthesis filter (104).

Abstract translation: 提供了一种音频编码装置，用于通过扩展层来校正核心层中具有不足的编码能力的部件。 在该装置中，核心层编码单元（101）对音频信号进行编码，扩展层编码单元（150）编码核心层编码单元（101）的编码残差，特征校正反向滤波器（102） LPC合成滤波器（104）的阶段对核心层中的编码能力不足的分量进行逆特性校正处理，并且在LPC合成滤波器（104）的后级配置的特征校正滤波器（105） 执行从LPC合成滤波器（104）输入的合成信号的特性校正处理。

公开(公告)号：US20090276210A1

公开(公告)日：2009-11-05

申请号：US12295073

申请日：2007-03-29

Applicant: Michiyo Goto ,  Koji Yoshida

Inventor： Michiyo Goto ,  Koji Yoshida

IPC: G10L19/14

CPC classification number: G10L19/008 ,  H04S1/00

Abstract: Disclosed is a stereo speech decoding device and others capable of reducing a stereo speech encoding bit rate and suppressing degradation of speech quality. In this device, a section 0 where only an L-channel signal SL(n) exists is identified, a monaural signal of the section 0 transmitted from the stereo speech encoding side is made to be an L-channel signal of section 0 SL(0)(n), and the L-channel signal SL(0)(n) of the section 0 is scale-adjusted so as to predict an R-channel signal SR(1)(n) of a section 1. A contribution of the R-channel signal SR(1)(n) of the predicted section 1 is subtracted from the monaural signal of the section 1 so as to isolate the L-channel signal SL(1)(n) of the section 1. This device continuously repeats the aforementioned scale adjustment and isolation process so as to obtain the L-channel signal SL(n) and the R-channel signal SR(n) of all the sections.

Abstract translation: 公开了一种能够降低立体声语音编码比特率并抑制语音质量降低的立体声语音解码装置等。 在该装置中，识别仅存在L信道信号SL（n）的部分0，将从立体声语音编码侧发送的部分0的单声道信号设为部分0 SL的L信道信号 0）（n），部分0的L信道信号SL（0）（n）被缩放调整，以便预测部分1的R信道信号SR（1）（n）。贡献 从部分1的单声道信号中减去预测部分1的R信道信号SR（1）（n），以隔离部分1的L信道信号SL（1）（n）。 设备连续地重复前述的缩放调整和隔离处理，以便获得所有部分的L信道信号SL（n）和R信道信号SR（n）。

公开(公告)号：US20090262945A1

公开(公告)日：2009-10-22

申请号：US12064995

申请日：2006-08-30

Applicant: Chun Woei Teo ,  Sua Hong Neo ,  Koji Yoshida ,  Michiyo Goto

Inventor： Chun Woei Teo ,  Sua Hong Neo ,  Koji Yoshida ,  Michiyo Goto

IPC: H04H20/88

CPC classification number: G10L19/24 ,  G10L19/008

Abstract: There is disclosed a stereo encoding device capable of accurately encoding a stereo signal at a low bit rate and suppressing delay in audio communication. The device performs monaural encoding in its first layer (110). In a second layer (120), a filtering unit (103) generates an LPC (Linear Predictive Coding) coefficient and generates a left channel drive sound source signal. A time region evaluation unit (104) and a frequency region evaluation unit (105) perform signal evaluation and prediction in both of their regions. A residual encoding unit (106) encodes a residual signal. A bit distribution control unit (107) adaptively distributes bits to the time region evaluation unit (104), the frequency region evaluation unit (105), and the residual encoding unit (106) according to a condition of the audio signal.

Abstract translation: 公开了一种能够以低比特率精确地编码立体声信号并抑制音频通信中的延迟的立体声编码装置。 设备在其第一层（110）中执行单声道编码。 在第二层（120）中，滤波单元（103）产生LPC（线性预测编码）系数并产生左声道驱动声源信号。 时间区域评估单元（104）和频率区域评估单元（105）在它们的两个区域中执行信号评估和预测。 残差编码单元（106）对剩余信号进行编码。 根据音频信号的条件，位分配控制单元（107）自适应地将时间区域评估单元（104），频率区域评估单元（105）和残差编码单元（106）分配。