公开(公告)号：US20100027618A1

公开(公告)日：2010-02-04

申请号：US12518347

申请日：2007-11-27

Applicant: Kazunori Ozawa

Inventor： Kazunori Ozawa

IPC: H04N11/04

CPC classification number: H04L1/08 ,  H04L1/0009 ,  H04L1/0041 ,  H04L1/0045 ,  H04L1/0075 ,  H04N7/17318 ,  H04N19/115 ,  H04N19/164 ,  H04N19/61 ,  H04N21/6375 ,  H04N21/8106

Abstract: This invention comprise a connection call processing unit for exchanging settings of encoded data and redundant data stored in each packet, and a setting of at least any one of encoding bit rates of a sound encoding unit and a redundant data generating unit, between the media transmitting/receiving apparatus, and a sound decoding unit for decoding at least any one of the encoded data and the redundant data based on the setting of the encoding bit rate exchanged between the media transmitting/receiving apparatuses, wherein the encoded data and the redundant data are separated from each packet based on the setting of the data stored in the packets exchanged between the media transmitting/receiving apparatuses, and lost encoded data is compensated with the redundant data, and thereby degradation of the media quality is prevented even when the packet loss occurs in transferring media encoded data in the packets via an IP network.

Abstract translation: 本发明包括一个连接呼叫处理单元，用于交换存储在每个分组中的编码数据和冗余数据的设置，以及在媒体发送之间设置声音编码单元和冗余数据生成单元的编码比特率中的至少一个 声音解码单元，用于基于在媒体发送/接收设备之间交换的编码比特率的设置来解码编码数据和冗余数据中的至少一个，其中编码数据和冗余数据是 基于存储在媒体发送/接收装置之间交换的分组中的数据的设置，与每个分组分离，丢失的编码数据被冗余数据补偿，从而即使发生分组丢失也可以防止媒体质量的劣化 在通过IP网络传送分组中的媒体编码数据时。

公开(公告)号：US07653251B2

公开(公告)日：2010-01-26

申请号：US11208886

申请日：2005-08-23

Applicant: Hiroaki Dei ,  Kazuhiro Koyama ,  Kazunori Ozawa

Inventor： Hiroaki Dei ,  Kazuhiro Koyama ,  Kazunori Ozawa

IPC: G06K9/36

CPC classification number: H04N7/152 ,  H04N21/23424 ,  H04N21/234309 ,  H04N21/234354 ,  H04N21/234363 ,  H04N21/234381 ,  H04N21/2662 ,  H04N21/632 ,  H04N21/6377

Abstract: A conversion server and a plurality of clients are connected via a transmission line. The conversion server that receives image coded data from the clients converts the image coding system in accordance with the coding system available to each client, the coding setting, and the status of the transmission line, and transmits the converted image coded data. For a client where the number of images that can be displayed is limited, the conversion server decodes a selected plurality items of image coded data, re-encodes the selected pieces into one composite image, and transmits the re-encoded image. The conversion server also comprises decoding processing units, one for each connected client, for decoding image coded data from each client. In response to a display image switching request from a client, the conversion server intraframe-codes the decoded image data of an image that will be used after the switching and transmits the intraframe-coded data. This enables an image to be switched quickly independently of the intraframe time of received image coded data that will be used after the switching.

Abstract translation: 转换服务器和多个客户端经由传输线连接。 从客户端接收图像编码数据的转换服务器根据可用于每个客户端的编码系统，编码设置和传输线路的状态来转换图像编码系统，并发送转换的图像编码数据。 对于可以显示的图像数量有限的客户端，转换服务器对所选择的多个图像编码数据进行解码，将所选择的片段重新编码为一个合成图像，并发送重新编码的图像。 转换服务器还包括解码处理单元，每个连接的客户机一个，用于对来自每个客户端的图像编码数据进行解码。 响应于来自客户端的显示图像切换请求，转换服务器对切换后将使用的图像的解码图像数据进行帧内编码，并发送帧内编码数据。 这使得可以独立于在切换之后将使用的接收图像编码数据的帧内时间来快速切换图像。

公开(公告)号：US07499853B2

公开(公告)日：2009-03-03

申请号：US11641009

申请日：2006-12-19

Applicant: Koji Yoshida ,  Hiroyuki Ehara ,  Masahiro Serizawa ,  Kazunori Ozawa

Inventor： Koji Yoshida ,  Hiroyuki Ehara ,  Masahiro Serizawa ,  Kazunori Ozawa

IPC: G10L19/12

CPC classification number: G10L19/005 ,  G10L25/12

Abstract: 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.

Abstract translation: 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。

公开(公告)号：US07298295B2

公开(公告)日：2007-11-20

申请号：US11443692

申请日：2006-05-31

Applicant: Kazunori Ozawa ,  Hiroaki Dei ,  Atsushi Hatabu

Inventor： Kazunori Ozawa ,  Hiroaki Dei ,  Atsushi Hatabu

IPC: H03M7/00

CPC classification number: G10L19/005 ,  G10L19/00 ,  G10L19/008 ,  G10L19/24 ,  H04L1/0045 ,  H04L1/0061 ,  H04L1/08

Abstract: A system and a method of suppressing outstanding degradation of decoded audio quality due to a transmission error of audio coded data are provided without feedback information from a receiver, thereby reducing the increase of the number of necessary transmission bands and the arithmetic complexity on the receiving side. A code conversion and transmission apparatus 100 for inputting audio coded data includes first to N-th code conversion and transmission units 102 and 104 to 106 for converting audio data to N pieces of coded data, and transmitting the data at predetermined or adaptively variable time intervals to M transmission lines 130. The second to N-th audio code conversion and transmission units 104 to 106 codes a frame at a compression rate equal to or higher than the rate of input coded data. The code conversion and reception apparatus 120 selects a transmission line using a selection unit 107, and selecting data from correctly received coded data in a frame or packet unit, thereby reconfiguring the coded data using a coded data reconfiguration unit 112.

公开(公告)号：US20070242663A1

公开(公告)日：2007-10-18

申请号：US11783657

申请日：2007-04-11

Applicant: Tatsuya Nakazawa ,  Kazunori Ozawa

Inventor： Tatsuya Nakazawa ,  Kazunori Ozawa

IPC: H04L12/66

CPC classification number: H04L29/06027 ,  H04L65/607 ,  H04L65/80

Abstract: When media communication is relayed through a circuit switching network and a packet switching network, packet congestion and packet loss is inhibited by limiting the bandwidth and the number of packets on the packet switching network. Audio packets received from the packet switching network are stored in a buffer for absorbing fluctuation on the packet switching network, and it is determined whether an encoded audio stream is acquirable from the buffer. If no acquisition, an encoded audio stream representing either silence or noise information is generated and multiplexed, and the multiplexed data is output to the circuit switching network. Multiplexed data constituted by a plurality of multiplexed encoded streams received from the circuit switching network is separated into respective encoded streams, and only encoded audio streams representing sound or noise information are packetized based on the frame information for the encoded audio streams and sent to the packet switching network.

Abstract translation: 当通过电路交换网络和分组交换网络中继媒体通信时，通过限制分组交换网络上的带宽和分组数量来阻止分组拥塞和分组丢失。 从分组交换网络接收的音频分组存储在用于吸收分组交换网络上的波动的缓冲器中，并且确定编码的音频流是否可从缓冲器获取。 如果没有获取，则生成并复用表示静音或噪声信息的编码音频流，并将多路复用数据输出到电路交换网络。 由从电路交换网络接收的多个多路复用编码流构成的复用数据被分离成各自的编码流，并且仅基于编码音频流的帧信息将代表声音或噪声信息的编码音频流分组化并发送到分组 交换网络。

公开(公告)号：US20070140664A1

公开(公告)日：2007-06-21

申请号：US10583080

申请日：2004-10-27

Applicant: Kazunori Ozawa

Inventor： Kazunori Ozawa

IPC: H04N7/26

CPC classification number: H04N19/577 ,  H04N19/117 ,  H04N19/137 ,  H04N19/172 ,  H04N19/176 ,  H04N19/44 ,  H04N19/61

Abstract: An apparatus and a method in which, even in case the transmitted speed on a transmission channel is low and hence it is not possible to provide a sufficient number of frames, a receiving side is able to restore a picture, which has not been received, with the use of a characteristic parameter, in such a manner that motion can be expressed sufficiently satisfactorily on the receiving side to improve the quality of a moving picture. The apparatus includes a decoder 200 for receiving a bitstream, obtained on compression/encoding of a moving picture, and for decoding a picture image from the bitstream; a characteristic parameter extractor 210 for extracting a characteristic parameter from a picture image restored, a moving picture reconstructor 212 for carrying out preset processing, using one or both of a temporally past characteristic parameter and a temporally future characteristic parameter, for restoring a picture image which has not been received.

Abstract translation: 即使在传输信道上的传输速度低的情况下，由于不可能提供足够数量的帧的情况下，接收侧也能恢复未被接收到的图像， 通过使用特征参数，可以在接收侧充分令人满意地表达运动，以提高运动画面的质量。 该装置包括：解码器200，用于接收通过压缩/编码运动图像获得的比特流，并用于从比特流解码图像图像; 用于从恢复的图像图像提取特征参数的特征参数提取器210，用于执行预设处理的运动图像重构器212，使用时间上过去的特征参数和时间上将来的特征参数中的一个或两个来恢复图像图像， 尚未收到

公开(公告)号：US20070127437A1

公开(公告)日：2007-06-07

申请号：US10576156

申请日：2004-09-24

Applicant: Kazunori Ozawa

Inventor： Kazunori Ozawa

IPC: H04L12/66

CPC classification number: H04L47/263 ,  H04L47/10 ,  H04L47/14 ,  H04L47/30

Abstract: In the transmission and reception of a medium signal, a device in accordance with the present invention reduces a medium signal deterioration generated by a data loss, caused by a bandwidth fluctuation in a wired IP network or a wireless network or by a wireless handover, and minimizes an increase in the amount of processing that is required for the signal deterioration. In a bidirectional medium transmission and reception, the reception side has a control unit 108 that monitors a storage amount of a buffer 111 in which a medium signal obtained by decoding a stream by a decoder 112 is stored, and transmits a control signal to a transmission line if the storage amount of the buffer exceeds or falls below a predetermined threshold, and the transmission side has a control unit 102 that encodes a medium signal and outputs a stream and, if a control signal is received from the transmission line, changes the compression rate of an encoder 104.

Abstract translation: 在媒体信号的发送和接收中，根据本发明的装置减少了由有线IP网络或无线网络中的带宽波动或无线切换引起的数据丢失产生的媒体信号恶化，以及 最小化信号恶化所需的处理量的增加。 在双向媒体发送和接收中，接收侧具有控制单元108，该控制单元108监视通过解码器112解码流所获得的媒体信号的缓冲器111的存储量，并将控制信号发送到传输 如果缓冲器的存储量超过或低于预定阈值，并且发送侧具有对媒体信号进行编码并输出流的控制单元102，并且如果从传输线接收到控制信号则改变压缩 编码器104的速率。

公开(公告)号：US07024354B2

公开(公告)日：2006-04-04

申请号：US09985853

申请日：2001-11-06

Applicant: Kazunori Ozawa

Inventor： Kazunori Ozawa

IPC: G10L19/04 ,  G10L19/08 ,  G10L19/10 ,  G10L19/14

CPC classification number: G10L19/083 ,  G10L19/06 ,  G10L2019/0012

Abstract: In response to a coded speech signal output from a speech coder, a speech decoder decodes the coded speech signal into a reproduction speech signal. If the reproduction speech signal meets predetermined conditions, for example, “silence”, “unvoiced sound”, and the like, the speech decoder further operates as the following. The speech decoder calculates spectral parameters based on the reproduction speech signal, and calculates an excitation signal on the basis of the reproduction speech signal and the spectral parameters. In the calculation, a level of the excitation signal is also obtained. The speech decoder smoothes in time at least one of the spectral parameters and the level of the excitation signal. The speech decoder synthesizes the excitation signal by using the synthesis filter constructed with the spectrum parameters, so as to reproduce the speech signal. The speech signal has an excellent quality even if a bit rate is low.

公开(公告)号：US06865534B1

公开(公告)日：2005-03-08

申请号：US09719826

申请日：1999-06-15

Applicant: Atsushi Murashima ,  Kazunori Ozawa

Inventor： Atsushi Murashima ,  Kazunori Ozawa

IPC: G10L19/04 ,  G10L11/00 ,  G10L13/02 ,  G10L19/00 ,  G10L19/06 ,  G10L19/12 ,  H03M7/30

CPC classification number: G10L19/12 ,  G10L19/0204 ,  G10L19/0212

Abstract: It is an object of the invention to excellently code a speech and music signal over all of bands in a speech and music signal code decoding apparatus having a band divided constitution. In order to achieve the object, a residue vector is generated by using an inverse filter (230 of FIG. 3) from a difference vector outputted from a first differencer (180 of FIG. 3). A band selecting circuit (250 of FIG. 3) generates n pieces of subvectors by using a component included in an arbitrary band in the residue vector subjected to orthogonal transformation. An orthogonal transformation coefficient quantizing circuit (260 of FIG. 3) quantizes n pieces of the subvectors.

Abstract translation: 本发明的目的是在具有带分割结构的语音和音乐信号编码解码装置中对所有频带上的语音和音乐信号进行卓越的编码。 为了实现该目的，通过使用从第一差分器（图3的180）输出的差矢量使用逆滤波器（图3的230）来生成残差矢量。 频带选择电路（图3的250）通过使用经过正交变换的残差矢量中的任意频带中包含的分量来生成n个子矢量。 正交变换系数量化电路（图3的260）对n个子矢量进行量化。

公开(公告)号：US06581031B1

公开(公告)日：2003-06-17

申请号：US09450305

申请日：1999-11-29

Applicant: Hironori Ito ,  Kazunori Ozawa ,  Masahiro Serizawa

Inventor： Hironori Ito ,  Kazunori Ozawa ,  Masahiro Serizawa

IPC: G10L1912

CPC classification number: G10L19/08 ,  G10L19/09

Abstract: In this speech encoding system, the limiter circuit is input with the delay of adaptive codebook obtained for the previous subframe, and the pitch cycle search range is limited so that the delay of adaptive codebook obtained for the previous subframe is not discontinuous to the delay of adaptive codebook to be obtained for the current subframe, and the pitch cycle search range limited is output to the pitch calculation circuit. The pitch calculation circuit is input with output signal Xw(n) of the perceptual weighting circuit and the pitch cycle search range output from the limiter, calculating the pitch cycle Top, then outputting at least one pitch cycle Top to the adaptive codebook circuit. The adaptive codebook circuit is input with the perceptual weighting signal x′w(n), the past excitation signal v(n) output from the gain quantization circuit, the perceptual weighting impulse response hw(n) output from the impulse response calculation circuit, and the pitch cycle Top from the pitch calculation circuit, searching near the pitch cycle, calculating the delay of adaptive codebook. With the above composition, the delay of adaptive codebook obtained for each subframe can be prevented from being discontinuous in the process of time.

Abstract translation: 在该语音编码系统中，限制电路以前一子帧获得的自适应码本的延迟输入，并且音调周期搜索范围被限制，使得对于前一个子帧获得的自适应码本的延迟不延迟 对于当前子帧获得的自适应码本，并且将音调周期搜索范围限制输出到音调计算电路。 音调计算电路输入感知加权电路的输出信号Xw（n）和从限制器输出的音调周期搜索范围，计算音调周期Top，然后将至少一个音调周期Top输出到自适应码本电路。 自适应码本电路输入感知加权信号x'w（n），从增益量化电路输出的过去激励信号v（n），从脉冲响应计算电路输出的感知加权脉冲响应hw（n） 和音调周期Top从音调计算电路，在音调周期附近搜索，计算自适应码本的延迟。 利用上述构成，可以防止在每个子帧中获得的自适应码本的延迟在时间过程中不连续。