公开(公告)号：US20100063828A1

公开(公告)日：2010-03-11

申请号：US12519248

申请日：2008-10-16

申请人： Tomokazu Ishikawa ,  Takeshi Norimatsu ,  Takashi Katayama

发明人： Tomokazu Ishikawa ,  Takeshi Norimatsu ,  Takashi Katayama

IPC分类号： G10L19/00

CPC分类号： H04M3/568 ,  G10L19/008 ,  G10L25/12 ,  H04L65/607

摘要： To provide an enhanced true-to-life atmosphere enjoyed in multipoint connecting, and reduce a calculation load at a multipoint connection unit, as well.A stream synthesizing device includes an input unit which inputs at least two coded signals each including a first downmix acoustic signal and an extended signal, each of first downmix acoustic signals being obtained by coding an acoustic signal into which at least two sound signals are downmixed, and the extended signal being for obtaining the at least two sound signals out of the first downmix acoustic signal; a coded signal generating unit which generates: a second downmix acoustic signal and an extended signal based on each of coded signals inputted by the input unit, the second downmix acoustic signal being for obtaining each of the first downmix acoustic signals, and the generated extended signal being for obtaining each of the first downmix acoustic signals out of the second downmix acoustic signal; and generate a coded signal including the generated second downmix acoustic signal, the generated extended signal, and each of extended signals included in the corresponding inputted coded signal; and an output unit which outputs the generated coded signal.

摘要翻译： 为了提供在多点连接中享受的增强的真实生活氛围，并减少多点连接单元的计算负荷。 流合成装置包括：输入单元，其输入至少两个编码信号，每个编码信号包括第一下混音声信号和扩展信号，每个第一下混音声信号是通过对至少两个声音信号进行混合的声信号进行编码而获得的， 并且所述扩展信号用于从所述第一下混声音信号中获得所述至少两个声音信号; 编码信号生成单元，其基于由所述输入单元输入的每个编码信号生成第二下混声音信号和扩展信号，所述第二下混声音信号用于获得所述第一下混声音信号中的每一个，并且所生成的扩展信号 用于从所述第二下混声音信号中获得所述第一下混声音信号中的每一个; 并生成包括所生成的第二下混声音信号，所生成的扩展信号和包括在相应的输入编码信号中的每个扩展信号的编码信号; 以及输出单元，其输出所生成的编码信号。

公开(公告)号：US08391513B2

公开(公告)日：2013-03-05

申请号：US12519248

申请日：2008-10-16

申请人： Tomokazu Ishikawa ,  Takeshi Norimatsu ,  Takashi Katayama

发明人： Tomokazu Ishikawa ,  Takeshi Norimatsu ,  Takashi Katayama

IPC分类号： H03G5/00

CPC分类号： H04M3/568 ,  G10L19/008 ,  G10L25/12 ,  H04L65/607

摘要： To provide an enhanced true-to-life atmosphere enjoyed in multipoint connecting, and reduce a calculation load at a multipoint connection unit, as well.A stream synthesizing device includes an input unit which inputs at least two coded signals each including a first downmix acoustic signal and an extended signal, each of first downmix acoustic signals being obtained by coding an acoustic signal into which at least two sound signals are downmixed, and the extended signal being for obtaining the at least two sound signals out of the first downmix acoustic signal; a coded signal generating unit which generates: a second downmix acoustic signal and an extended signal based on each of coded signals inputted by the input unit, the second downmix acoustic signal being for obtaining each of the first downmix acoustic signals, and the generated extended signal being for obtaining each of the first downmix acoustic signals out of the second downmix acoustic signal; and generate a coded signal including the generated second downmix acoustic signal, the generated extended signal, and each of extended signals included in the corresponding inputted coded signal; and an output unit which outputs the generated coded signal.

摘要翻译： 为了提供在多点连接中享受的增强的真实生活氛围，并减少多点连接单元的计算负荷。 流合成装置包括：输入单元，其输入至少两个编码信号，每个编码信号包括第一下混音声信号和扩展信号，每个第一下混音声信号是通过对至少两个声音信号进行混合的声信号进行编码而获得的， 并且所述扩展信号用于从所述第一下混声音信号中获得所述至少两个声音信号; 编码信号生成单元，其基于由所述输入单元输入的每个编码信号生成第二下混声音信号和扩展信号，所述第二下混声音信号用于获得所述第一下混声音信号中的每一个，并且所生成的扩展信号 用于从所述第二下混声音信号中获得所述第一下混声音信号中的每一个; 并生成包括所生成的第二下混声音信号，所生成的扩展信号和包括在相应的输入编码信号中的每个扩展信号的编码信号; 以及输出单元，其输出所生成的编码信号。

公开(公告)号：US20140058737A1

公开(公告)日：2014-02-27

申请号：US13996644

申请日：2012-10-24

申请人： Tomokazu Ishikawa ,  Takeshi Norimatsu ,  Kok Seng Chong ,  Dan Zhao

发明人： Tomokazu Ishikawa ,  Takeshi Norimatsu ,  Kok Seng Chong ,  Dan Zhao

IPC分类号： G10L19/002

CPC分类号： G10L19/002 ,  G10L19/02 ,  G10L19/12 ,  G10L19/20

摘要： A hybrid sound signal decoder decodes a bitstream including audio frames encoded by an audio encoding process using a low delay filter bank and speech frames encoded by a speech encoding process using linear prediction coefficients. When a current frame to be decoded is an ith frame which is an initial speech frame after switching from an audio frame to a speech frame, the hybrid sound signal decoder generates sub-frames which are a signal corresponding to an i−1th frame before being encoded, using a sub-frame which is a signal generated using a signal of the i−1th frame before being encoded, the signal of the i−1th frame being obtained by decoding the ith frame.

摘要翻译： 混合声音信号解码器使用低延迟滤波器组和使用线性预测系数的语音编码处理编码的语音帧来解码包括由音频编码处理编码的音频帧的比特流。 当要解码的当前帧是从音频帧切换到语音帧之后的初始语音帧的第i帧时，混合声音信号解码器产生作为与第i-1帧对应的信号的子帧 使用作为在编码之前使用第i-1帧的信号产生的信号的子帧，通过解码第i帧获得第i-1帧的信号。

公开(公告)号：US20110268279A1

公开(公告)日：2011-11-03

申请号：US13141169

申请日：2010-10-21

申请人： Tomokazu Ishikawa ,  Takeshi Norimatsu ,  Kok Seng Chong ,  Huan Zhou ,  Haishan Zhong

发明人： Tomokazu Ishikawa ,  Takeshi Norimatsu ,  Kok Seng Chong ,  Huan Zhou ,  Haishan Zhong

IPC分类号： H04R5/00

CPC分类号： G10L19/265 ,  G10L19/008 ,  G10L19/0212 ,  G10L19/09 ,  G10L25/90

摘要： Provided is an encoding device (1) including: a pitch contour analysis unit (101) which detects information, a dynamic time-warping unit (102) which generates, based on the information, pitch change ratios (Tw_ratio in FIG. 18) within a range (86) including a range (86a) of the pitch change ratios corresponding to absolute pitch differences of 42 cents or larger; a first lossless coding unit (103) which codes the generated pitch parameters (102x); a time-warping unit (104) which shifts a pitch of a signal according to the information; and a second encoding unit which codes a signal (104x) obtained by the shifting.

摘要翻译： 提供了一种编码装置（1），包括：检测信息的音调轮廓分析单元（101），动态时间扭曲单元（102），其基于该信息生成内部的音调变化率（图18中的Tw_ratio） 包括对应于42分或以上的绝对音调差的音调变化率的范围（86a）的范围（86） 编码所生成的音调参数（102x）的第一无损编码单元（103）。 时间扭曲单元（104），其根据该信息移动信号的音调; 以及第二编码单元，其对通过移位获得的信号（104x）进行编码。

公开(公告)号：US20100198589A1

公开(公告)日：2010-08-05

申请号：US12679814

申请日：2009-07-28

申请人： Tomokazu Ishikawa ,  Takeshi Norimatsu ,  Kok Seng Chong ,  Huan Zhou

发明人： Tomokazu Ishikawa ,  Takeshi Norimatsu ,  Kok Seng Chong ,  Huan Zhou

IPC分类号： G10L19/00

CPC分类号： G10L19/008

摘要： The delay in a multi-channel audio coding apparatus and a multi-channel audio decoding apparatus is reduced. The audio coding apparatus includes: a downmix signal generating unit (410) that generates, in a time domain, a first downmix signal that is one of a 1-channel audio signal and a 2-channel audio signal from an input multi-channel audio signal; a downmix signal coding unit (404) that codes the first downmix signal; a first t-f converting unit (401) that converts the input multi-channel audio signal into a multi-channel audio signal in a frequency domain; and a spatial information calculating unit (409) that generates spatial information for generating a multi-channel audio signal from a downmix signal.

摘要翻译： 减少了多声道音频编码装置和多声道音频解码装置中的延迟。 音频编码装置包括：下混合信号生成单元，其在时域中生成作为来自输入多声道音频的1声道音频信号和2声道音频信号之一的第1降混信号 信号; 编码第一下混信号的下混信号编码单元（404） 第一t-f转换单元，其将输入的多声道音频信号转换成频域中的多声道音频信号; 以及空间信息计算单元（409），其生成用于从降混信号生成多声道音频信号的空间信息。

公开(公告)号：US06484142B1

公开(公告)日：2002-11-19

申请号：US09551796

申请日：2000-04-18

申请人： Shuji Miyasaka ,  Takeshi Norimatsu ,  Mineo Tsushima ,  Tomokazu Ishikawa ,  Yoshiaki Sawada

发明人： Shuji Miyasaka ,  Takeshi Norimatsu ,  Mineo Tsushima ,  Tomokazu Ishikawa ,  Yoshiaki Sawada

IPC分类号： G10L1914

CPC分类号： H03M7/42

摘要： An encoder of the present invention includes: G storage sections for storing G groups of data; a selection section for selecting one of H Huffman codebooks having codebook numbers for each of the groups of data; G encoding sections Huffman-encoding the G groups of data by using the selected Huffman codebook; and an encoding section for encoding the codebook number of each Huffman codebook selected. The selection section includes a calculation section for calculating a code length and a control section for selecting one of the Huffman codebooks. When the Huffman codebook selected is an unsigned codebook, a number of bits required for sign information has previously been added to the calculated code length.

摘要翻译： 本发明的编码器包括：G存储部分，用于存储G组数据; 选择部分，用于选择具有每组数据的码本号码的霍夫曼代码簿之一; G编码部分通过使用所选的霍夫曼码本对G组数据进行霍夫曼编码; 以及编码部，用于编码所选择的每个霍夫曼码本的码本号。 选择部分包括用于计算代码长度的计算部分和用于选择霍夫曼码本之一的控制部分。 当选择的霍夫曼代码簿是无符号码本时，已经将计算的代码长度添加到符号信息所需的位数。

公开(公告)号：US09093080B2

公开(公告)日：2015-07-28

申请号：US13389276

申请日：2011-06-06

申请人： Tomokazu Ishikawa ,  Takeshi Norimatsu ,  Huan Zhou ,  Kok Seng Chong ,  Haishan Zhong

发明人： Tomokazu Ishikawa ,  Takeshi Norimatsu ,  Huan Zhou ,  Kok Seng Chong ,  Haishan Zhong

IPC分类号： G10L19/00 ,  G10L21/038 ,  G10L21/04

CPC分类号： G10L19/02 ,  G10L21/038 ,  G10L21/04

摘要： To provide a bandwidth extension method which allows reduction of computation amount in bandwidth extension and suppression of deterioration of quality in the bandwidth to be extended. In the bandwidth extension method: a low frequency bandwidth signal is transformed into a QMF domain to generate a first low frequency QMF spectrum; pitch-shifted signals are generated by applying different shifting factors on the low frequency bandwidth signal; a high frequency QMF spectrum is generated by time-stretching the pitch-shifted signals in the QMF domain; the high frequency QMF spectrum is modified; and the modified high frequency QMF spectrum is combined with the first low frequency QMF spectrum.

摘要翻译： 提供一种允许减少带宽扩展中的计算量并抑制要扩展的带宽中的质量劣化的带宽扩展方法。 在带宽扩展方法中：将低频带宽信号变换成QMF域以产生第一低频QMF频谱; 通过在低频带宽信号上应用不同的移位因子来生成音调移位信号; 通过对QMF域中的音调移位信号进行时间扩展来产生高频QMF频谱; 改变高频QMF频谱; 并且修改的高频QMF频谱与第一低频QMF频谱组合。

公开(公告)号：US20120136670A1

公开(公告)日：2012-05-31

申请号：US13389276

申请日：2011-06-06

申请人： Tomokazu Ishikawa ,  Takeshi Norimatsu ,  Huan Zhou ,  Kok Seng Chong ,  Haishan Zhong

发明人： Tomokazu Ishikawa ,  Takeshi Norimatsu ,  Huan Zhou ,  Kok Seng Chong ,  Haishan Zhong

IPC分类号： G10L19/00

CPC分类号： G10L19/02 ,  G10L21/038 ,  G10L21/04

摘要： To provide a bandwidth extension method which allows reduction of computation amount in bandwidth extension and suppression of deterioration of quality in the bandwidth to be extended. In the bandwidth extension method: a low frequency bandwidth signal is transformed into a QMF domain to generate a first low frequency QMF spectrum; pitch-shifted signals are generated by applying different shifting factors on the low frequency bandwidth signal; a high frequency QMF spectrum is generated by time-stretching the pitch-shifted signals in the QMF domain; the high frequency QMF spectrum is modified; and the modified high frequency QMF spectrum is combined with the first low frequency QMF spectrum.

摘要翻译： 提供一种允许减少带宽扩展中的计算量并抑制要扩展的带宽中的质量劣化的带宽扩展方法。 在带宽扩展方法中：将低频带宽信号变换成QMF域以产生第一低频QMF频谱; 通过在低频带宽信号上应用不同的移位因子来生成音调移位信号; 通过对QMF域中的音调移位信号进行时间扩展来产生高频QMF频谱; 改变高频QMF频谱; 并且修改的高频QMF频谱与第一低频QMF频谱组合。

公开(公告)号：US20120022676A1

公开(公告)日：2012-01-26

申请号：US13256055

申请日：2010-10-19

申请人： Tomokazu Ishikawa ,  Takeshi Norimatsu ,  Kok Seng Chong ,  Huan Zhou ,  Haishan Zhong

发明人： Tomokazu Ishikawa ,  Takeshi Norimatsu ,  Kok Seng Chong ,  Huan Zhou ,  Haishan Zhong

IPC分类号： G06F17/00

CPC分类号： G10L21/04 ,  G10L21/0388

摘要： To provide an audio signal processing apparatus which can perform, with low operation amount, audio signal processing that is either time stretch and/or compression processing or frequency modulation processing. The audio signal processing apparatus is intended to transform an input audio signal sequence using a predetermined adjustment factor. The audio signal processing apparatus includes a filter bank (2601) which transforms the input audio signal sequence into Quadrature Mirror Filter (QMF) coefficients using a filter for Quadrature Mirror Filter analysis (a QMF analysis filter) and an adjusting unit (2602) which adjusts the QMF coefficients based on a predetermined adjustment factor.

摘要翻译： 提供一种音频信号处理装置，其可以以低操作量执行时间延伸和/或压缩处理或频率调制处理的音频信号处理。 音频信号处理装置旨在使用预定的调节因子来变换输入音频信号序列。 音频信号处理装置包括滤波器组（2601），其使用用于正交镜滤波器分析的滤波器（QMF分析滤波器）和调整单元（2602）将输入音频信号序列变换为正交镜像滤波器（QMF）系数，调整单元（2602） 基于预定调整因子的QMF系数。

公开(公告)号：US08019614B2

公开(公告)日：2011-09-13

申请号：US12065378

申请日：2006-08-31

申请人： Yoshiaki Takagi ,  Kok Seng Chong ,  Takeshi Norimatsu ,  Shuji Miyasaka ,  Akihisa Kawamura ,  Kojiro Ono ,  Tomokazu Ishikawa

发明人： Yoshiaki Takagi ,  Kok Seng Chong ,  Takeshi Norimatsu ,  Shuji Miyasaka ,  Akihisa Kawamura ,  Kojiro Ono ,  Tomokazu Ishikawa

IPC分类号： G10L19/00

CPC分类号： G10L19/26 ,  G10L19/008 ,  G10L19/0204 ,  H04S2420/03

摘要： A temporal processing apparatus includes: a splitter splitting an audio signal, included in the sub-band domain, into diffuse signals indicating reverberating components and direct signals indicating non-reverberating components; a downmix unit generating a downmix signal by downmixing the direct signals; BPFs respectively generating a bandpass downmix signal and bandpass diffuse signals; normalization processing units respectively generating a normalized downmix signal and normalized diffuse signals; a scale computation processing unit computing, on a predetermined time slot basis, a scale factor indicating the magnitude of energy of the normalized downmix signal with respect to energy of the normalized diffuse signals; a calculating unit generating scale diffuse signals; a HPF generating high-pass diffuse signals; an adding unit generating addition signals; and a synthesis filter bank performing synthesis filter processing on the addition signals and transforming the addition signals into the time domains.

摘要翻译： 时间处理装置包括：分离器，将包括在子带域中的音频信号分成指示混响分量的漫射信号和指示非混响分量的直接信号; 下混合单元，通过使直接信号下混合来产生降混信号; BPF分别产生带通下混信号和带通漫射信号; 归一化处理单元，分别产生归一化的下混信号和归一化扩散信号; 比例计算处理单元在预定时隙的基础上计算指示归一化的下混信号相对于归一化扩散信号的能量的能量的大小的比例因子; 计算单元，生成缩放漫射信号; HPF产生高通漫反射信号; 添加单元生成附加信号; 以及合成滤波器组，对加法信号执行合成滤波处理，并将加法信号转换成时域。