公开(公告)号：US06501863B1

公开(公告)日：2002-12-31

申请号：US09162120

申请日：1998-09-28

申请人： Takahiro Fukuhara ,  Seiji Kimura ,  Kenzo Akagiri

发明人： Takahiro Fukuhara ,  Seiji Kimura ,  Kenzo Akagiri

IPC分类号： G06K936

CPC分类号： G06T9/007

摘要： The present invention relates to an image coding apparatus/method, an image decoding apparatus/method, and a transmission medium, which conduct high-efficiency coding or decoding of an image, and applicable in a system performing efficient transmission or accumulation of images. In coding an image after splitting the image into a plurality of bands, the invention is characterized in a process comprising the steps of splitting the image into a plurality of bands, applying quadrature transformation to the images of the resultant band components, quantizing the resultant quadrature transformation coefficients, and generating a coded bit stream by variable-length-coding the quantization coefficient. This permits coding less susceptible to block strain at a high compression ratio.

摘要翻译： 图像编码装置/方法，图像解码装置/方法以及传输介质技术领域本发明涉及对图像进行高效率编码或解码的图像编码装置/方法，图像解码装置/方法以及传输介质，可应用于执行图像的有效传输或累积的系统。 在将图像分割成多个频带之后对图像进行编码时，本发明的特征在于包括以下步骤的步骤：将图像分割成多个频带，对所得到的频带分量的图像应用正交变换，量化所得到的正交 变换系数，并且通过对量化系数进行可变长度编码来生成编码比特流。 这允许在高压缩比下对块应变不太敏感的编码。

公开(公告)号：US20050193012A1

公开(公告)日：2005-09-01

申请号：US10965269

申请日：2004-10-14

申请人： Shinako Matsuyama ,  Kenzo Akagiri ,  Koji Suginuma

发明人： Shinako Matsuyama ,  Kenzo Akagiri ,  Koji Suginuma

IPC分类号： G06Q50/26 ,  G06F7/00 ,  G06F17/30 ,  G06F21/60 ,  G06Q50/00 ,  G06Q99/00 ,  G09C1/00 ,  H04L9/32

CPC分类号： G06F21/6245 ,  G06F2221/2129 ,  G06Q30/02

摘要： Disclosed is a private information management apparatus in which the information of high practical utility for a user in person may be acquired more efficiently and with higher responsiveness than is possible with a technique of keyword retrieval over the network. To this end, a private information management apparatus 1 for acquiring and storing the experience information pertinent to an event experienced by a user, and the private information input by a user, includes, as a structure for authenticating a counterpart terminal, with which the user is to exchange the experience information and the private information, an authentication key generating unit 59 for generating an authentication key for authenticating the terminal for which the information is publicized, an encryption unit 60 for encrypting part of the experience information and the private information stored as the publicized information, based on the encryption key, and an authentication processing unit 61 for verifying whether or not the authentication key received from the external terminal is one from the designated terminal. In this manner, the user is able to publicize only the information the user may feel not disinclined to publicize to another user, based on the taste information stored from user to user.

摘要翻译： 公开了一种私人信息管理装置，其中可以比通过网络进行关键字检索的技术更有效地并且以更高的响应性获得用于用户的高实用性的信息。 为此，用于获取和存储与用户经历的事件相关的体验信息和用户输入的私人信息的私人信息管理装置1包括用于认证对方终端的结构，用户 用于交换体验信息和私人信息的认证密钥生成单元59，用于生成用于认证公开该信息的终端的认证密钥的加密单元60，用于加密体验信息的一部分的加密单元60和存储为 基于加密密钥的公开信息，以及用于验证从外部终端接收的认证密钥是否为来自指定终端的认证密钥的认证处理单元61。 以这种方式，用户能够基于从用户到用户存储的嗜好信息，仅公开用户可能不想不愿意向另一用户公开的信息。

公开(公告)号：US06794996B2

公开(公告)日：2004-09-21

申请号：US10239848

申请日：2003-04-01

申请人： Kyoya Tsutsui ,  Kenzo Akagiri ,  Tadao Yoshida ,  Naoya Haneda

发明人： Kyoya Tsutsui ,  Kenzo Akagiri ,  Tadao Yoshida ,  Naoya Haneda

IPC分类号： H03M700

CPC分类号： H04N19/124 ,  G06Q20/12 ,  G06Q20/123 ,  G06Q20/1235 ,  G06Q30/06 ,  G07F17/0014 ,  G11B20/00086 ,  G11B20/00818 ,  G11B20/00978 ,  G11B20/10 ,  G11B27/034 ,  G11B27/105 ,  G11B27/3063 ,  G11B2220/20 ,  G11B2220/216 ,  G11B2220/2525 ,  G11B2220/2545 ,  G11B2220/41 ,  H04N19/18 ,  H04N19/60 ,  H04N21/234345 ,  H04N21/234354 ,  H04N21/23476 ,  H04N21/4621 ,  H04N21/4627 ,  H04N21/47211 ,  H04N21/8113 ,  H04N21/8549

摘要： A center (content supply center) (1865) for storing and managing contents, and user terminals (1861 to 1864) used by respective users are connected via networks (861 to 867). The center (1865) sends signals of contents for trial viewing/listening to the user terminals for free or at a low price. The user terminals (1861 to 1864) can receive the signals of the contents for trial viewing/listening and select and purchase only the content that the user likes from the received contents so as to reproduce the content with high quality. This enables trial viewing/listening. It is not necessary to download a large quantity of data with high quality from the content supply center or the like and there will be more opportunities of content purchase.

摘要翻译： 用于存储和管理内容的中心（内容供应中心）（1865）和各用户使用的用户终端（1861至1864）经由网络（861至867）连接。 中心（1865）免费或以低价格发送用于试听/收听用户终端的内容信号。 用户终端（1861至1864）可以接收用于试听/收听的内容的信号，并仅从所接收的内容中仅选择和购买用户喜欢的内容，以便以高质量再现内容。 这样可以进行试听/聆听。 没有必要从内容供应中心等下载大量高质量的数据，并且将有更多的内容购买机会。

公开(公告)号：USRE36559E

公开(公告)日：2000-02-08

申请号：US245451

申请日：1994-05-18

申请人： Yoshihito Fujiwara ,  Tomoko Umezawa ,  Masayuki Nishiguchi ,  Makoto Akune ,  Naoto Iwahashi ,  Kenzo Akagiri

发明人： Yoshihito Fujiwara ,  Tomoko Umezawa ,  Masayuki Nishiguchi ,  Makoto Akune ,  Naoto Iwahashi ,  Kenzo Akagiri

IPC分类号： H04B1/66 ,  H03M7/00

CPC分类号： H04B1/667 ,  H04B1/665

摘要： This invention relates to a digital signal encoding apparatus in which the .[.width of the range in.]. .Iadd.digital signal is divided into frequency components in plural frequency bands and the bandwidth of the frequency bands is .Iaddend.selected to be wider for .[.the.]. higher .[.frequency range.]. .Iadd.frequencies .Iaddend.of the digital .[.signals divided into a plurality of regions.]. .Iadd.signal .Iaddend.and in which the encoded signals are synthesized for the respective .[.ranges.]. .Iadd.frequency bands .Iaddend.wherein encoding is controlled as a function of the output detecting the characteristics of the .Iadd.frequency .Iaddend.components .[.of.]. .Iadd.in .Iaddend.the .[.divided.]. frequency .Iadd.bands .Iaddend.and the detection time interval is selected to be longer for the lower frequency .Iadd.bands .Iaddend.to enable more efficient encoding to be performed as a function of the properties of input digital signals.

摘要翻译： 数字信号编码装置本发明涉及一种数字信号编码装置，其中[数字信号的范围]被划分为多个频带的频率分量，频带的带宽被选择为较宽[频率范围 ]频率的数字[信号被分成多个区域]信号，并且其中编码的信号被合成为各个[频带]频带，其中编码被控制为输出的函数，检测频率分量[ ]，并且对于较低频带选择检测时间间隔更长，以使得能够根据输入数字信号的性质执行更有效的编码。

公开(公告)号：US5873065A

公开(公告)日：1999-02-16

申请号：US446689

申请日：1995-11-20

申请人： Kenzo Akagiri ,  Mark Franklin Davis ,  Craig Campbell Todd ,  Ray Milton Dolby

发明人： Kenzo Akagiri ,  Mark Franklin Davis ,  Craig Campbell Todd ,  Ray Milton Dolby

IPC分类号： G11B20/10 ,  G10L19/00 ,  G10L19/008 ,  G10L19/02 ,  G11B20/00 ,  H03M7/30 ,  H03M7/40 ,  H04B1/66 ,  H04B14/04 ,  H04H5/00 ,  H04H20/88 ,  H04S1/00 ,  H04S3/00

CPC分类号： H04H20/88 ,  G11B20/00007 ,  H04B1/665 ,  H04S1/007 ,  H04S3/00 ,  G11B2020/10592 ,  G11B2020/10601

摘要： A multi-channel signal compressor for compressing digital sound signals in the respective channels of a multi-channel sound system. The apparatus comprises a first-stage compression system and a second-stage compression system. In the first-stage compression system, a coupling circuit performs coupling between the digital sound signals of at least two of the channels to generate coupling-processed signals, one for each of the channels. A compressor circuit receives the coupling-processed signals from the coupling circuit and frequency divides each coupling-processed signal into frequency range signals in respective frequency ranges, and compresses the frequency range signals obtained by dividing each coupling-processed signal to generate a first-stage compressed signal. In the second-stage compression system, a determining circuit receives the first-stage compressed signal for each channel from the first-stage compression system and determines an energy for each channel from the first-stage compressed signal of the respective channel. A channel bit apportionment decision circuit operates in response to the determining circuit, and apportions a predetermined number of bits among the channels to apportion a number of bits to each channel. Finally, an additional compressor additionally compresses the first-stage compressed signal of each channel using, for each channel, the number of bits apportioned to the respective channel by the channel bit apportionment decision circuit.

摘要翻译： PCT No.PCT / US94 / 14267 Sec。 371日期：1995年11月20日 102（e）1995年11月20日日期PCT 1994年12月7日提交的用于在多声道声音系统的各个声道中压缩数字声音信号的多声道信号压缩器。 该装置包括第一级压缩系统和第二级压缩系统。 在第一级压缩系统中，耦合电路在至少两个通道的数字声音信号之间执行耦合，以产生耦合处理的信号，每个通道一个信号。 压缩机电路从耦合电路接收经耦合处理的信号，并且将每个经耦合处理的信号频率分成各个频率范围内的频率范围信号，并且压缩通过划分每个耦合处理的信号而获得的频率范围信号，以产生第一级 压缩信号。 在第二级压缩系统中，确定电路从第一级压缩系统接收每个通道的第一级压缩信号，并从相应通道的第一级压缩信号确定每个通道的能量。 通道位分配判定电路响应于确定电路而工作，并且在通道中分配预定数量的位以将每个通道的位数分配。 最后，额外的压缩器另外使用信道比特分配决定电路对每个信道压缩每个信道的第一级压缩信号，以分配给相应信道的比特数。

公开(公告)号：US5757938A

公开(公告)日：1998-05-26

申请号：US439089

申请日：1995-05-08

申请人： Kenzo Akagiri ,  Makoto Akune

发明人： Kenzo Akagiri ,  Makoto Akune

IPC分类号： G10L21/02 ,  G10L19/00 ,  H03G7/00 ,  H03M7/30 ,  H04B1/66 ,  H04B14/04 ,  H04R3/04 ,  H03G5/00

CPC分类号： H03G7/007 ,  H04B1/665

摘要： An apparatus and method for processing an input digital signal in which an input digital audio signal at an input terminal 11 is supplied via a one-block delay unit 17 to a difference unit 20, an output of which is processed by a noise-shaping circuit 21 and restored by a near-instantaneous compressor 26, a near-instantaneous expander 27 and an accumulator 28 so as to be fed back to the difference unit 20 to find a difference between a signal of a directly preceding sample and a signal of the current sample. The noise shaping circuit 21 modifies the spectrum of the quantization noise by taking aural characteristics, such as equal-loudness characteristics or masking characteristics, into account for diminishing the level of the quantization noise as perceived by the ear. The noise level as perceived may be reduced in a manner desirable for the sound quality without changing the existing format.

摘要翻译： 一种用于处理输入数字信号的装置和方法，其中输入端子11处的输入数字音频信号经由一块延迟单元17提供给差分单元20，差分单元20的输出由噪声整形电路 21并由近瞬时压缩机26，近瞬时扩展器27和累加器28恢复，以反馈到差分单元20，以找到直接在前的采样的信号与电流的信号之间的差 样品。 噪声整形电路21通过采用诸如等响度特性或掩蔽特性之类的听觉特性来改变量化噪声的频谱，以减少由耳朵感知的量化噪声的电平。 可以以不改变现有格式的声音质量的方式来减少感知到的噪声水平。

公开(公告)号：US5754427A

公开(公告)日：1998-05-19

申请号：US662059

申请日：1996-06-12

申请人： Kenzo Akagiri

发明人： Kenzo Akagiri

IPC分类号： G11B20/10 ,  G10L19/00 ,  G10L19/02 ,  G11B20/00 ,  G11B20/12 ,  H03M7/36 ,  H04B1/66 ,  H04B14/04 ,  H03M7/30

CPC分类号： G11B20/00007 ,  H04B1/665 ,  H04B14/046

摘要： In a compact disc having a fixed bit rate, the bit rate is rendered substantially variable for improving the sound quality. Each sample is represented by a fixed length of 16 bits. For a sample with a bit surplus K, its data is substantially represented by upper 14 bits. For the lower two bits of the block K are allocated data of lower two bits of data of a bit deficit block L substantially represented by 18 bits.

摘要翻译： 在具有固定比特率的光盘中，使得比特率基本上是可变的，以改善声音质量。 每个样本由16位的固定长度表示。 对于具有多余K的样本，其数据基本上由高14位表示。 对于块K的低两位分配基本上由18位表示的位缺失块L的低两位数据的数据。

公开(公告)号：US5530750A

公开(公告)日：1996-06-25

申请号：US198863

申请日：1994-02-18

申请人： Kenzo Akagiri

发明人： Kenzo Akagiri

IPC分类号： G11B20/00 ,  G11B20/10 ,  G11B27/031 ,  G11B27/30 ,  H03M7/30 ,  H03M7/50 ,  H04B1/66 ,  H04L9/00

CPC分类号： G11B27/3063 ,  G11B20/00007 ,  G11B20/10527 ,  G11B27/031 ,  H04B1/665

摘要： An apparatus, method, and system in which a compressed digital signal is derived from a digital input signal by compressing the digital input signal in a selected one of plural compression modes. The digital output signal has a different bit rate in each compression mode, but the digital input signal is received at the same bit rate in all the compression modes. In the method, spectral coefficients are derived from the digital input signal, and are grouped by frequency and by time into bands. Each band has a frequency width, and the frequency width of at least one of the bands is set according to the selected compression mode. The spectral coefficients in each band are quantized to provide quantized spectral coefficients. Finally, the quantized spectral coefficients in each band are included in the compressed digital signal, together with sub information for each band.

摘要翻译： 一种装置，方法和系统，其中通过以多种压缩模式中选择的一种压缩数字输入信号，从数字输入信号导出压缩数字信号。 在每个压缩模式下，数字输出信号具有不同的比特率，但在所有压缩模式下，数字输入信号都以相同的比特率被接收。 在该方法中，频谱系数从数字输入信号导出，并且按频率和时间分组成频带。 每个频带具有频率宽度，并且根据所选择的压缩模式来设置至少一个频带的频率宽度。 量化每个频带中的频谱系数以提供量化频谱系数。 最后，每个频带中的量化频谱系数与每个频带的子信息一起包括在压缩数字信号中。

公开(公告)号：US5438643A

公开(公告)日：1995-08-01

申请号：US230303

申请日：1994-04-20

申请人： Kenzo Akagiri ,  Yoshiaki Oikawa ,  Kyoya Tsutsui

发明人： Kenzo Akagiri ,  Yoshiaki Oikawa ,  Kyoya Tsutsui

IPC分类号： G11B20/00 ,  G11B20/10 ,  G11B27/031 ,  G11B27/30 ,  G11B27/34 ,  G10L3/02

CPC分类号： G11B20/10527 ,  G11B20/00007 ,  G11B27/031 ,  G11B27/3063 ,  G11B27/34 ,  G11B2220/2525 ,  G11B2220/2545 ,  G11B2220/61

摘要： Apparatus for recording an information signal as bit-compressed digital data. The apparatus comprises a bit compressor and a recording system. The bit compressor receives the information signal as digital data. The digital data include bits, and the bits include redundant bits. The bit compressor compresses the digital data by removing only the redundant bits from the digital data to generate the bit-compressed digital data at a variable bit rate. The redundant bits are bits that result in a quantizing noise level lower than an allowed noise level at which the quantizing noise is imperceptible to a human auditory sense. The allowable noise level is determined by a masking threshold and a minimum audibility limit. The recording system receives the bit-compressed digital data from the bit compressor and records the bit-compressed data on a recording medium.

摘要翻译： 用于将信息信号记录为位压缩的数字数据的装置。 该装置包括位压缩器和记录系统。 位压缩器接收信息信号作为数字数据。 数字数据包括位，并且这些位包括冗余位。 位压缩器通过从数字数据中仅去除冗余位来压缩数字数据，以可变比特率生成位压缩的数字数据。 冗余位是导致量化噪声电平低于量化噪声对于人类听觉无法察觉的允许噪声电平的位。 允许的噪声电平由屏蔽阈值和最小可听限度决定。 记录系统从位压缩器接收位压缩的数字数据，并将位压缩的数据记录在记录介质上。

公开(公告)号：US5396577A

公开(公告)日：1995-03-07

申请号：US994113

申请日：1992-12-22

申请人： Yoshiaki Oikawa ,  Kenzo Akagiri

发明人： Yoshiaki Oikawa ,  Kenzo Akagiri

IPC分类号： G10L13/04 ,  G10L13/08 ,  G10L5/00

CPC分类号： G10L13/08 ,  G10L13/047

摘要： In a speech synthesizing apparatus, importance degree information indicative of a degree of importance with respect to each text portion of input original text data is added to this text portion. Then, the original text data with such importance degree information is input. When a rapid reading process, or a head searching process is carried out for the original text input, speech synthesis is carried out by controlling several stages which text portion should be skipped, or at which speed, the text portions should be synthesized, in response to a speed instruction and importance degree information which are being input into the speech synthesizing apparatus.

摘要翻译： 在语音合成装置中，向该文本部分添加表示相对于输入原始文本数据的每个文本部分的重要程度的重要程度信息。 然后，输入具有这种重要度信息的原始文本数据。 当对原始文本输入进行快速读取处理或头部搜索处理时，通过控制几个阶段来执行语音合成，该阶段应当跳过文本部分，或者应该以什么速度合成文本部分 涉及正在输入到语音合成装置中的速度指令和重要度信息。