专利快速检索-快速检索全球专利，免费商用专利数据库-IPRDB

1. 发明申请

US20110051940A1 DECODING DEVICE, CODING AND DECODING DEVICE, AND DECODING METHOD 有权
标题翻译：解码设备，编码和解码设备以及解码方法
公开(公告)号：US20110051940A1
公开(公告)日：2011-03-03
申请号：US12920141
申请日：2010-03-26
申请人： Tomokazu Ishikawa , Takeshi Norimatsu , Huan Zhou , Zhong Hai Shan , Kok Seng Chong
发明人： Tomokazu Ishikawa , Takeshi Norimatsu , Huan Zhou , Zhong Hai Shan , Kok Seng Chong
IPC分类号： H04R5/00
CPC分类号： H04L12/40013 , H04M3/56 , H04R5/04
摘要： There was no method of positioning virtual sound sources of object signals obtained from received coded object information, in a listening space on a receiving site side. Provided are an object decoding unit (806) that receives a plurality of pieces of coded acoustic information coded on an object basis and decodes the received coded acoustic information on an object basis to generate object signals, and an a rendering unit (807) that positions, for each acoustic signal resulting from synthesizing the object signals, a virtual sound source of each of the object signals resulting from the object-based decoding, in a listening space, and priority information indicating a priority of the acoustic signal is multiplexed with the coded acoustic information, and the rendering unit (807) determines, with reference to the priority indicated in the priority information or the number of object signals included in the acoustic signal, a size of an area of the listening space which is allocated to the acoustic signal, and positions, within the area, the virtual sound source of each of the object signals included in the acoustic signal.
摘要翻译：在接收站点侧的收听空间中没有定位从接收到的编码对象信息获得的对象信号的虚拟声源的方法。提供了一种对象解码单元（806），其接收基于对象编码的多条编码声音信息，并且基于对象对接收到的编码声音信息进行解码以生成对象信号;以及渲染单元（807），其定位对于从合成对象信号得到的每个声信号，在收听空间中从基于对象的解码得到的每个对象信号的虚拟声源和指示声信号的优先级的优先级信息被多路复用声音信息，并且再现单元（807）参照优先权信息中所指示的优先级或包括在声学信号中的对象信号的数量来确定分配给声信号的收听空间的面积的大小，以及在区域内的位置，包括在声信号中的每个对象信号的虚拟声源。

2. 发明授权

US08718285B2 Decoding device, coding and decoding device, and decoding method 有权
标题翻译：解码装置，编码和解码装置以及解码方法
公开(公告)号：US08718285B2
公开(公告)日：2014-05-06
申请号：US12920141
申请日：2010-03-26
申请人： Tomokazu Ishikawa , Takeshi Norimatsu , Huan Zhou , Zhong Hai Shan , Kok Seng Chong
发明人： Tomokazu Ishikawa , Takeshi Norimatsu , Huan Zhou , Zhong Hai Shan , Kok Seng Chong
IPC分类号： H04R5/00
CPC分类号： H04L12/40013 , H04M3/56 , H04R5/04
摘要： An object decoding unit receives a plurality of pieces of coded acoustic information coded on an object basis and decodes the received coded acoustic information on an object basis to generate object signals. A rendering unit positions, for each acoustic signal resulting from synthesizing the object signals, a virtual sound source of each of the object signals resulting from the object-based decoding, in a listening space, and priority information indicating a priority of the acoustic signal is multiplexed with the coded acoustic information. The rendering unit determines, with reference to the priority indicated in the priority information or the number of object signals included in the acoustic signal, a size of an area of the listening space which is allocated to the acoustic signal, and positions, within the area, the virtual sound source of each of the object signals included in the acoustic signal.
摘要翻译：对象解码单元接收基于对象编码的多条编码声音信息，并且基于对象对接收到的编码声音信息进行解码以生成对象信号。渲染单元对于从合成对象信号得到的每个声信号，在收听空间中对基于对象的解码产生的每个对象信号的虚拟声源和指示声信号的优先级的优先级信息进行定位，与编码的声学信息复用。渲染单元参考优先权信息中所指示的优先级或包括在声信号中的对象信号的数量来确定分配给声信号的收听空间的面积的大小，以及在该区域内的位置，包括在声信号中的每个对象信号的虚拟声源。

3. 发明授权

US08504184B2 Combination device, telecommunication system, and combining method 有权
标题翻译：组合设备，电信系统及组合方法
公开(公告)号：US08504184B2
公开(公告)日：2013-08-06
申请号：US12935797
申请日：2010-02-04
申请人： Tomokazu Ishikawa , Takeshi Norimatsu , Huan Zhou , Zhong Hai Shan , Kok Seng Chong
发明人： Tomokazu Ishikawa , Takeshi Norimatsu , Huan Zhou , Zhong Hai Shan , Kok Seng Chong
IPC分类号： G06F17/00
CPC分类号： G10L19/008 , H04M3/56
摘要： A combination device (305) according to the present invention includes: a detection unit (501) that detects active coded bitstreams that are effective coded bitstreams from a plurality of coded bitstreams (116) within a predetermined time period; a first combining unit (504) that combines, from a plurality of downmix sub-streams (115) included in the coded bitstreams (116), only downmix sub-streams (115) included in the active coded bitstreams so as to generate a combined downmix sub-stream (121); and a second combining unit (506) that combines, from a plurality of parameter sub-streams (113) included in the coded bitstreams (116), only parameter sub-streams (113) included In the active coded bitstreams so as to generate a combined parameter sub-stream (122).
摘要翻译：根据本发明的组合装置（305）包括：检测单元（501），其在预定时间段内检测来自多个编码比特流（116）的有效编码比特流的有效编码比特流; 从包括在编码比特流（116）中的多个下混子流（115）组合的第一组合单元（504）仅将包括在有源编码比特流中的子流（115）进行下混合，以便生成组合下混合子流（121）; 以及第二组合单元（506），其从包括在编码比特流（116）中的多个参数子流（113）组合仅包括在所述有源编码比特流中的参数子流（113），以便生成组合参数子流（122）。

4. 发明申请

US20110029113A1 COMBINATION DEVICE, TELECOMMUNICATION SYSTEM, AND COMBINING METHOD 有权
标题翻译：组合设备，电信系统和组合方法
公开(公告)号：US20110029113A1
公开(公告)日：2011-02-03
申请号：US12935797
申请日：2010-02-04
申请人： Tomokazu Ishikawa , Takeshi Norimatsu , Huan Zhou , Zhong Hai Shan , Kok Seng Chong
发明人： Tomokazu Ishikawa , Takeshi Norimatsu , Huan Zhou , Zhong Hai Shan , Kok Seng Chong
IPC分类号： G06F17/00
CPC分类号： G10L19/008 , H04M3/56
摘要： A combination device (305) according to the present invention includes: a detection unit (501) that detects active coded bitstreams that are effective coded bitstreams from a plurality of coded bitstreams (116) within a predetermined time period; a first combining unit (504) that combines, from a plurality of downmix sub-streams (115) included in the coded bitstreams (116), only downmix sub-streams (115) included in the active coded bitstreams so as to generate a combined downmix sub-stream (121); and a second combining unit (506) that combines, from a plurality of parameter sub-streams (113) included in the coded bitstreams (116), only parameter sub-streams (113) included In the active coded bitstreams so as to generate a combined parameter sub-stream (122).
摘要翻译：根据本发明的组合装置（305）包括：检测单元（501），其在预定时间段内检测来自多个编码比特流（116）的有效编码比特流的有效编码比特流; 从包括在编码比特流（116）中的多个下混子流（115）组合的第一组合单元（504）仅将包括在有源编码比特流中的子流（115）进行下混合，以便生成组合下混合子流（121）; 以及第二组合单元（506），其从包括在编码比特流（116）中的多个参数子流（113）组合仅包括在所述有源编码比特流中的参数子流（113），以便生成组合参数子流（122）。

5. 发明申请

US20110268279A1 AUDIO ENCODING DEVICE, DECODING DEVICE, METHOD, CIRCUIT, AND PROGRAM 有权
标题翻译：音频编码设备，解码设备，方法，电路和程序
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。

6. 发明申请

US20100198589A1 AUDIO CODING APPARATUS, AUDIO DECODING APPARATUS, AUDIO CODING AND DECODING APPARATUS, AND TELECONFERENCING SYSTEM 有权
标题翻译：音频编码设备，音频解码设备，音频编码和解码设备以及电话系统
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。

7. 发明授权

US09093080B2 Bandwidth extension method, bandwidth extension apparatus, program, integrated circuit, and audio decoding apparatus 有权
标题翻译：带宽扩展方法，带宽扩展装置，程序，集成电路和音频解码装置
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。

8. 发明申请

US20120136670A1 BANDWIDTH EXTENSION METHOD, BANDWIDTH EXTENSION APPARATUS, PROGRAM, INTEGRATED CIRCUIT, AND AUDIO DECODING APPARATUS 有权
标题翻译：带宽扩展方法，带宽扩展设备，程序，集成电路和音频解码设备
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。

9. 发明申请

US20120022676A1 AUDIO SIGNAL PROCESSING APPARATUS, AUDIO CODING APPARATUS, AND AUDIO DECODING APPARATUS 有权
标题翻译：音频信号处理装置，音频编码装置和音频解码装置
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。

10. 发明授权

US08886548B2 Audio encoding device, decoding device, method, circuit, and program 有权
标题翻译：音频编码装置，解码装置，方法，电路和程序
公开(公告)号：US08886548B2
公开(公告)日：2014-11-11
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： G10L21/04 , G10L19/008 , H04S3/02 , G10L19/26 , G10L19/09 , G10L19/02
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。

你已经成功收藏专利！

检索式保存成功!

IPRDB

热门服务

关于我们

友情链接

联系方式