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41. 发明授权

US08271272B2 Scalable encoding device, scalable decoding device, and method thereof 有权
标题翻译：可扩展编码装置，可扩展解码装置及其方法
公开(公告)号：US08271272B2
公开(公告)日：2012-09-18
申请号：US11587379
申请日：2005-04-19
申请人： Hiroyuki Ehara , Koji Yoshida
发明人： Hiroyuki Ehara , Koji Yoshida
IPC分类号： G10L19/00
CPC分类号： G10L19/07 , G10L19/24
摘要： There is disclosed a scalable encoding device capable of increasing the conversion performance from a narrow-band LSP to a wide-band LSP (prediction accuracy when predicting the wide-band LSP from the narrow-band LSP) and realizing a high-performance band scalable LSP encoding. The device includes a conversion coefficient calculation unit (109) for calculating a conversion coefficient by using a narrow-band quantization LSP which has been outputted from a narrow-band LSP encoding unit (103) and a wide-band quantization LSP which has been outputted from a wide-band LSP encoding unit (107). The wide-band LSP encoding unit (107) multiplies the narrow-band quantization LSP with the conversion coefficient inputted from the conversion coefficient calculation unit (109) so as to convert it into a wide-band LSP. The wide-band LSP is multiplied by a weight coefficient to calculate a prediction wide-band LSP. The wide-band LSP encoding unit (107) encodes an error signal between the obtained prediction wide-band LSP and the wide-band LSP so as to obtain a wide-band quantization LSP.
摘要翻译：公开了一种能够提高从窄带LSP到宽带LSP的转换性能的可伸缩编码装置（预测来自窄带LSP的宽带LSP的预测精度），实现高性能频带可伸缩性 LSP编码。该装置包括转换系数计算单元（109），用于通过使用从窄带LSP编码单元（103）输出的窄带量化LSP和已经输出的宽带量化LSP来计算转换系数来自宽带LSP编码单元（107）。宽带LSP编码单元（107）将窄带量化LSP与从转换系数计算单元（109）输入的转换系数相乘，以将其转换为宽带LSP。宽带LSP乘以权重系数以计算预测宽带LSP。宽带LSP编码单元（107）对获得的预测宽带LSP与宽带LSP之间的误差信号进行编码，以获得宽带量化LSP。

42. 发明授权

US08204745B2 Encoder, decoder, encoding method, and decoding method 有权
标题翻译：编码器，解码器，编码方法和解码方法
公开(公告)号：US08204745B2
公开(公告)日：2012-06-19
申请号：US13158944
申请日：2011-06-13
申请人： Masahiro Oshikiri , Hiroyuki Ehara , Koji Yoshida
发明人： Masahiro Oshikiri , Hiroyuki Ehara , Koji Yoshida
IPC分类号： G10L19/02
CPC分类号： G10L21/038
摘要： An encoding apparatus and method for generating low-frequency-band encoding information and high-frequency-band encoding information from an original signal. The encoding apparatus includes a first spectrum calculator that calculates a first spectrum of a low frequency band from a decoded signal of the low-frequency-band encoding information, a second spectrum calculator that calculates a second spectrum from the original signal, an estimator that divides a high frequency band of the second spectrum into a plurality of bands and estimates the second spectrum included in each band, using the first spectrum, and a first error component encoder that encodes a first error component between the high frequency band of the second spectrum and an estimated spectrum. A corresponding decoding apparatus and method provides decoding.
摘要翻译：一种用于从原始信号产生低频带编码信息和高频带编码信息的编码装置和方法。编码装置包括：第一频谱计算器，根据低频带编码信息的解码信号计算低频带的第一频谱;第二频谱计算器，从原始信号计算第二频谱;估计器，其将将第二频谱的高频带转换成多个频带，并使用第一频谱估计包括在每个频带中的第二频谱，以及第一误差分量编码器，其对第二频谱的高频带和估计频谱。相应的解码装置和方法提供解码。

43. 发明授权

US07991611B2 Speech encoding apparatus and speech encoding method that encode speech signals in a scalable manner, and speech decoding apparatus and speech decoding method that decode scalable encoded signals 有权
标题翻译：以可缩放的方式编码语音信号的语音编码装置和语音编码方法，以及解码可缩放编码信号的语音解码装置和语音解码方法
公开(公告)号：US07991611B2
公开(公告)日：2011-08-02
申请号：US12089814
申请日：2006-10-13
申请人： Hiroyuki Ehara , Koji Yoshida
发明人： Hiroyuki Ehara , Koji Yoshida
IPC分类号： G10L19/14
CPC分类号： G10L19/24 , G10L19/26
摘要： An audio encoding device for correcting a component having insufficient encoding capability in a core layer by an extended layer. A core layer encoder encodes an audio signal. An extended layer encoder encodes an encoding residual of the core layer encoder. A characteristic correction inverse filter arranged at a pre-stage of an LPC synthesis filter subjects the component having insufficient encoding capability in the core layer to an inverse characteristic correction process, and a characteristic correction filter arranged at a post-stage of the LPC synthesis filter performs a process for characteristic correction of the synthesis signal inputted from the LPC synthesis filter.
摘要翻译：一种音频编码装置，用于通过扩展层校正在核心层中具有不足的编码能力的部件。核心层编码器对音频信号进行编码。扩展层编码器编码核心层编码器的编码残差。布置在LPC合成滤波器的前级的特征校正反相滤波器将核心层中具有不足的编码能力的分量进行逆特性校正处理，以及布置在LPC合成滤波器的后级的特征校正滤波器执行从LPC合成滤波器输入的合成信号的特性校正处理。

44. 发明授权

US07957962B2 Fixed codebook searching apparatus and fixed codebook searching method 有权
公开(公告)号：US07957962B2
公开(公告)日：2011-06-07
申请号：US12392880
申请日：2009-02-25
申请人： Hiroyuki Ehara , Koji Yoshida
发明人： Hiroyuki Ehara , Koji Yoshida
IPC分类号： G10L19/12
CPC分类号： G10L19/107
摘要： A fixed codebook searching apparatus which slightly suppresses an increase in the operation amount, even if the filter applied to the excitation pulse has the characteristic that it cannot be represented by a lower triangular matrix and realizes a quasi-optimal fixed codebook search. This fixed codebook searching apparatus is provided with an algebraic codebook that generates a pulse excitation vector; a convolution operation section that convolutes an impulse response of auditory weighted synthesis filter into an impulse response vector that has a value at negative times, to generate a second impulse response vector that has a value at second negative times; a matrix generating section that generates a Toeplitz-type convolution matrix by means of the second impulse response vector; and a convolution operation section that convolutes the matrix generated by matrix generating section into the pulse excitation vector generated by algebraic codebook.

45. 发明申请

US20110064229A1 AUDIO SIGNAL DECODING DEVICE AND BALANCE ADJUSTMENT METHOD FOR AUDIO SIGNAL DECODING DEVICE 有权
标题翻译：用于音频信号解码装置的音频信号解码装置和平衡调整方法
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。

46. 发明申请

US20090326930A1 SPEECH DECODING APPARATUS AND SPEECH ENCODING APPARATUS 有权
标题翻译：语音解码设备和语音编码设备
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。

47. 发明授权

US07499853B2 Speech decoder and code error compensation method 失效
标题翻译：语音解码器和码错误补偿方法
公开(公告)号：US07499853B2
公开(公告)日：2009-03-03
申请号：US11641009
申请日：2006-12-19
申请人： Koji Yoshida , Hiroyuki Ehara , Masahiro Serizawa , Kazunori Ozawa
发明人： Koji Yoshida , Hiroyuki Ehara , Masahiro Serizawa , Kazunori Ozawa
IPC分类号： G10L19/12
CPC分类号： G10L19/005 , G10L25/12
摘要： When an error is detected in coded data in the current frame, data separation section 201 separates the data into coding parameters first. Then, mode information decoding section 202 outputs decoding mode information in the previous frame and uses this as the mode information of the current frame. Furthermore, using the lag parameter code and gain parameter code of the current frame obtained at data separation section 201 and the mode information, lag parameter decoding section 204 and gain parameter decoding section 205 adaptively calculate a lag parameter and gain parameter to be used in the current frame according to the mode information.
摘要翻译：当在当前帧中的编码数据中检测到错误时，数据分离部201首先将数据分离为编码参数。然后，模式信息解码部分202输出前一帧中的解码模式信息，并将其用作当前帧的模式信息。此外，使用在数据分离部201获取的当前帧的滞后参数代码和增益参数码以及模式信息，滞后参数解码部204和增益参数解码部205自适应地计算要使用的滞后参数和增益参数当前帧根据模式信息。

48. 发明授权

US08452590B2 Fixed codebook searching apparatus and fixed codebook searching method 有权
标题翻译：固定码本搜索装置和固定码本搜索方法
公开(公告)号：US08452590B2
公开(公告)日：2013-05-28
申请号：US13093294
申请日：2011-04-25
申请人： Hiroyuki Ehara , Koji Yoshida
发明人： Hiroyuki Ehara , Koji Yoshida
IPC分类号： G10L19/22
CPC分类号： G10L19/107
摘要： A fixed codebook searching apparatus, includes a convolution operator, implemented by at least one processor, that convolves an impulse response of a perceptually weighted synthesis filter with an impulse response vector that has values at negative times, to generate a second impulse response vector that has values at negative times. A matrix generator, implemented by at least one processor, generates a Toeplitz-type convolution matrix using the second impulse response vector generated by the convolution operator. A searcher, implemented by at least one processor, performs a codebook search by maximizing a term using the Toeplitz-type convolution matrix.
摘要翻译：一种固定码本搜索装置，包括由至少一个处理器实现的卷积运算符，其将感知加权的合成滤波器的脉冲响应与具有负时间的值的脉冲响应向量卷积，以产生具有值在负时间。由至少一个处理器实现的矩阵生成器使用由卷积算子产生的第二脉冲响应向量生成Toeplitz型卷积矩阵。由至少一个处理器实现的搜索器通过使用Toeplitz型卷积矩阵最大化一个项来执行码本搜索。

49. 发明授权

US08255213B2 Speech decoding apparatus, speech encoding apparatus, and lost frame concealment method 有权
标题翻译：语音解码装置，语音编码装置和丢失帧隐藏方法
公开(公告)号：US08255213B2
公开(公告)日：2012-08-28
申请号：US12373085
申请日：2007-07-11
申请人： Koji Yoshida , Hiroyuki Ehara
发明人： Koji Yoshida , Hiroyuki Ehara
IPC分类号： G10L19/00
CPC分类号： G10L19/005
摘要： A sound decoding device is capable of improving the lost frame compensation performance and improving quality of the decoded sound. A rise frame sound source compensation unit 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 updates the average sound source pattern held in an average sound source pattern holding unit over a plurality of frames. When a frame is lost, an LPC synthesis unit performs LPC synthesis on a decoded sound source signal by using the compensation sound source signal inputted via a switching unit and a decoded LPC parameter from an LPC decoding unit and outputs the compensation decoded sound signal.
摘要翻译：声音解码装置能够改善丢失的帧补偿性能并提高解码声音的质量。上升帧声源补偿单元当当前帧是丢失帧和上升帧时产生补偿声源信号。平均声源图案更新单元通过多个帧来更新保持在平均声源图案保持单元中的平均声源图案。当帧丢失时，LPC合成单元通过使用经由切换单元输入的补偿声源信号和来自LPC解码单元的解码LPC参数对解码的声源信号执行LPC合成，并输出补偿解码声音信号。

50. 发明授权

US08229749B2 Wide-band encoding device, wide-band LSP prediction device, band scalable encoding device, wide-band encoding method 有权
标题翻译：宽带编码装置，宽带LSP预测装置，频带可伸缩编码装置，宽带编码方法
公开(公告)号：US08229749B2
公开(公告)日：2012-07-24
申请号：US11721358
申请日：2005-12-09
申请人： Hiroyuki Ehara , Koji Yoshida , Toshiyuki Morii
发明人： Hiroyuki Ehara , Koji Yoshida , Toshiyuki Morii
IPC分类号： G10L13/00 , G10L19/00 , G10L21/04
CPC分类号： G10L19/24 , G10L19/07 , G10L21/038
摘要： There is provided a wide-band LSP prediction device and others capable of predicting a wide-band LSP from a narrow-band LSP with a high quantization efficiency and a high accuracy while suppressing the size of a conversion table correlating the narrow-band LSP to the wide-band LSP. In this device, a non-linear prediction unit (102) performs non-linear prediction by using a converted wide-band LSP inputted from a narrow-band/wide-band conversion unit (101) and inputs the non-linear prediction result to an amplifier (103). The converted wide-band LSP is inputted to an amplifier (104). An adder (122) adds multiplication results (vectors) inputted from the amplifiers (103, 104).
摘要翻译：提供了一种宽带LSP预测装置和其他能够在抑制与窄带LSP相关联的转换表的大小的同时以高量化效率和高精度预测来自窄带LSP的宽带LSP的方法宽带LSP。在该装置中，非线性预测单元（102）通过使用从窄带/宽带转换单元（101）输入的转换宽带LSP进行非线性预测，并将非线性预测结果输入到放大器（103）。转换的宽带LSP被输入到放大器（104）。加法器（122）将从放大器（103,104）输入的相乘结果相加。

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