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1. 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。

2. 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。

3. 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。

4. 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。

5. 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。

6. 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。

7. 发明申请

US20130090929A1 HYBRID AUDIO ENCODER AND HYBRID AUDIO DECODER 有权
标题翻译：混合音频编码器和混合音频解码器
公开(公告)号：US20130090929A1
公开(公告)日：2013-04-11
申请号：US13703044
申请日：2011-06-14
申请人： Tomokazu Ishikawa , Takeshi Norimatsu , Haishan Zhong , Kok Seng Chong , Huan Zhou
发明人： Tomokazu Ishikawa , Takeshi Norimatsu , Haishan Zhong , Kok Seng Chong , Huan Zhou
IPC分类号： G10L19/04
CPC分类号： G10L19/04 , G10L19/0212 , G10L19/022 , G10L19/107 , G10L19/20
摘要： Provided are a new hybrid audio decoder and a new hybrid audio encoder having block switching for speech signals and audio signals. Currently, very low bitrate audio coding methods for speech and audio signal are proposed. These audio coding methods cause very long delay. Generally, in coding an audio signal, algorithm delay tends to be long to achieve higher frequency resolution. In coding a speech signal, the delay needs to be reduced because the speech signal is used for telecommunication. To balance fine coding quality for these two kinds of input signals with very low bitrate, this invention provides a combination of a low delay filter bank like AAC-ELD and a CELP coding method.
摘要翻译：提供了一种新的混合音频解码器和具有用于语音信号和音频信号的块切换的新的混合音频编码器。目前，提出了用于语音和音频信号的非常低的比特率音频编码方法。这些音频编码方法造成很长的延迟。通常，在对音频信号进行编码时，算法延迟往往很长以达到更高的频率分辨率。在对语音信号进行编码时，由于语音信号用于通信，所以需要减小延迟。为了平衡具有非常低的比特率的这两种输入信号的精细编码质量，本发明提供了像AAC-ELD这样的低延迟滤波器组和CELP编码方法的组合。

8. 发明授权

US08311810B2 Reduced delay spatial coding and decoding apparatus and teleconferencing system 有权
标题翻译：减少延迟空间编码和解码装置和电话会议系统
公开(公告)号：US08311810B2
公开(公告)日：2012-11-13
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： G10L21/00 , G10L19/02 , H04R5/00
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 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 that codes the first downmix signal; a first t-f converting unit that converts the input multi-channel audio signal into a multi-channel audio signal in a frequency domain; and a spatial information calculating unit 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：降混信号生成单元，其在时域中生成来自输入多声道音频信号的1声道音频信号和2声道音频信号之一的第1下混合信号; 下混合信号编码单元，对第一下混信号进行编码; 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，其生成用于从降混信号生成多声道音频信号的空间信息。

9. 发明申请

US20110182432A1 CODING APPARATUS AND DECODING APPARATUS 有权
标题翻译：编码装置和解码装置
公开(公告)号：US20110182432A1
公开(公告)日：2011-07-28
申请号：US13121991
申请日：2010-07-30
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： H04R5/00
CPC分类号： G10L19/008 , H04S2420/03
摘要： A coding apparatus which suppresses an extreme increase in a bit rate, includes: a downmixing and coding unit (301) that downmixes audio signals that have been provided, to reduce the number of channels to be fewer than the number of the provided audio signals, and to code the downmix signals; an object parameter extracting unit (304) that extracts parameters indicating correlation between the audio signals; and a multiplexing circuit (309) that multiplexes the extracted parameters with the generated downmix coded signals. The object parameter extracting unit (304) includes: an object classifying unit (305) that classifies each of the provided audio signals into a predetermined one of types based on audio characteristics; and an object parameter extracting circuit (308) that extracts parameters using a temporal granularity and a frequency granularity each of which is determined for a corresponding one of the types.
摘要翻译：抑制比特率的极大增加的编码装置包括：降混和编码单元（301），其降低已经提供的音频信号，以将频道数减少到少于所提供的音频信号的数量，并对下混合信号进行编码; 提取表示音频信号之间的相关性的参数的对象参数提取单元（304）和多路复用电路（309），将所提取的参数与生成的下混合编码信号进行复用。对象参数提取单元（304）包括：对象分类单元（305），其基于音频特性将所提供的音频信号分类为预定类型; 以及目标参数提取电路（308），其使用时间粒度和频率粒度提取参数，每个时间粒度和频率粒度被确定为相应的一种类型。

10. 发明授权

US09275650B2 Hybrid audio encoder and hybrid audio decoder which perform coding or decoding while switching between different codecs 有权
标题翻译：混合音频编码器和混合音频解码器，其在不同编解码器之间切换时执行编码或解码
公开(公告)号：US09275650B2
公开(公告)日：2016-03-01
申请号：US13703044
申请日：2011-06-14
申请人： Tomokazu Ishikawa , Takeshi Norimatsu , Haishan Zhong , Kok Seng Chong , Huan Zhou
发明人： Tomokazu Ishikawa , Takeshi Norimatsu , Haishan Zhong , Kok Seng Chong , Huan Zhou
IPC分类号： G10L19/02 , G10L19/04 , G10L19/107 , G10L19/20 , G10L19/022
CPC分类号： G10L19/04 , G10L19/0212 , G10L19/022 , G10L19/107 , G10L19/20
摘要： A new hybrid audio decoder and a new hybrid audio encoder having block switching for speech signals and audio signals are provided. Currently, very low bitrate audio coding methods for speech and audio signals are proposed. These audio coding methods cause very long delays. Generally, in coding an audio signal, an algorithm delay tends to be long to achieve higher frequency resolution. In coding a speech signal, the delay needs to be reduced because the speech signal is used for telecommunication. To balance fine coding quality for speech and audio input signals with very low bitrate, a combination of a low delay filter bank like AAC-ELD and a CELP coding method is provided.
摘要翻译：提供了一种新的混合音频解码器和具有用于语音信号和音频信号的块切换的新的混合音频编码器。目前，提出了用于语音和音频信号的非常低的比特率音频编码方法。这些音频编码方法造成很长的延迟。通常，在对音频信号进行编码时，算法延迟往往很长以达到更高的频率分辨率。在对语音信号进行编码时，由于语音信号用于通信，所以需要减小延迟。为了平衡具有非常低的比特率的语音和音频输入信号的精细编码质量，提供了诸如AAC-ELD的低延迟滤波器组和CELP编码方法的组合。

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