公开(公告)号：US08655649B2

公开(公告)日：2014-02-18

申请号：US13243015

申请日：2011-09-23

Applicant: Kosuke Tsujino ,  Kei Kikuiri ,  Nobuhiko Naka

Inventor： Kosuke Tsujino ,  Kei Kikuiri ,  Nobuhiko Naka

IPC: G10L19/00 ,  G10L21/00

CPC classification number: G10L19/03 ,  G10L19/0208 ,  G10L19/0212 ,  G10L19/06 ,  G10L19/167 ,  G10L19/24 ,  G10L19/26 ,  G10L21/038 ,  G10L21/04

Abstract: A linear prediction coefficient of a signal represented in a frequency domain is obtained by performing linear prediction analysis in a frequency direction by using a covariance method or an autocorrelation method. After the filter strength of the obtained linear prediction coefficient is adjusted, filtering may be performed in the frequency direction on the signal by using the adjusted coefficient, whereby the temporal envelope of the signal is shaped. This reduces the occurrence of pre-echo and post-echo and improves the subjective quality of the decoded signal, without significantly increasing the bit rate in a bandwidth extension technique in the frequency domain represented by spectral band replication.

Abstract translation: 通过使用协方差方法或自相关方法通过在频率方向上进行线性预测分析来获得在频域中表示的信号的线性预测系数。 在调整了所获得的线性预测系数的滤波强度之后，可以通过使用调整后的系数在频率方向上对信号进行滤波，由此信号的时间包络成形。 这减少了前回波和后回波的发生，并提高了解码信号的主观质量，而不会显着增加由频谱带复制表示的频域中的带宽扩展技术中的比特率。

公开(公告)号：US06744784B1

公开(公告)日：2004-06-01

申请号：US09214445

申请日：1998-12-29

Applicant: Nobuhiko Naka ,  Toshiro Kawahara

Inventor： Nobuhiko Naka ,  Toshiro Kawahara

IPC: H04J300

CPC classification number: H04J3/0605

Abstract: A variable length frame transmission method making it possible to accurately and easily establish synchronism at the receiver side without redundancy of system under an environment in which a code error easily occurs. In a transmitter, a variable length frame division section 1 divides a variable length frame f into code strings f1 and f2 having a length ratio of 1:1. A first synchronization flag addition section 3-1 adds a synchronization flag S1 to the head of the code string f1 and a second synchronization flag addition section 3-2 adds a synchronization flag S2 to the head of the code string f2. The synchronization flags have contents different from each other, but they have the same length. Code strings having synchronization flags are multiplexed by a changeover switch 4 and formed into a variable length frame. A series of variable length frames obtained from the changeover switch 4 are transmitted to a receiver as serial data. In the receiver, the start and end points of each frame is obtained based on the position of each synchronization flag in the serial data.

Abstract translation: 一种可变长度帧传输方法，使得可以在接收机侧准确和容易地建立同步，而在容易发生码错误的环境下，系统的冗余不可避免。在发射机中，可变长度帧分割部分1将可变长度帧 f变换成长度比为1:1的代码串f1和f2。 第一同步标记附加部分3-1将同步标记S1加到代码串f1的头部，第二同步标记相加部分3-2将同步标记S2加到代码串f2的头部。 同步标记具有彼此不同的内容，但是它们具有相同的长度。 具有同步标志的代码串由转换开关4复用并形成为可变长度的帧。 从切换开关4获得的一系列可变长度帧作为串行数据发送到接收机。 在接收机中，基于串行数据中的每个同步标志的位置来获得每帧的起点和终点。

公开(公告)号：US06714908B1

公开(公告)日：2004-03-30

申请号：US09462126

申请日：1999-12-29

Applicant: Nobuhiko Naka

Inventor： Nobuhiko Naka

IPC: G10L2102

CPC classification number: G10L19/12 ,  G10L19/005 ,  H03M7/425 ,  H03M7/6005

Abstract: A speech decoder 10 comprises a decoding processing portion 11 and an amplification process control portion 12. Here, the decoding processing potion 11 is a device for decoding a received coded speech signal (bitstream) BS and outputting a decoded speech signal SP. Additionally, the amplification process control portion 12 monitors the state of occurrence of frame errors in the coded speech signal BS, and when the number of successive frame errors exceeds a predetermined reference frame error number, outputs amplification instructions for a predetermined number of frames after the successive frame errors disappear. As a result, instead of codebook data DCB obtained by a decoding process of the decoding processing portion 11, amplified codebook data DACB are supplied to a synthesis filter portion 17, and is written into the codebook decoder 18 of the decoding processing portion 11 as new original codebook data DCBO. Therefore, after successive frame errors, it is possible to decode normally input coded speech signals BS at a power close to the originally intended power, thus enabling the subjective sound quality of the decoded speech SP to be improved.

Abstract translation: 语音解码器10包括解码处理部分11和放大处理控制部分12.这里，解码处理部分11是用于对接收到的编码语音信号（比特流）BS进行解码并输出解码语音信号SP的装置。 此外，放大处理控制部分12监视编码语音信号BS中的帧错误的发生状态，并且当连续帧错误的数量超过预定的参考帧错误号时，输出预定的帧数之后的放大指令 连续帧错误消失。 结果，代替通过解码处理部分11的解码处理获得的码本数据DCB，放大码本数据DACB被提供给合成滤波器部分17，并作为新的被写入解码处理部分11的码本解码器18 原始码本数据DCBO。 因此，在连续的帧错误之后，可以以接近原来的功率的功率对正常输入的编码语音信号BS进行解码，从而能够提高解码语音SP的主观音质。

公开(公告)号：US07505900B2

公开(公告)日：2009-03-17

申请号：US10500103

申请日：2002-12-25

Applicant: Kei Kikuiri ,  Nobuhiko Naka ,  Tomoyuki Ohya

Inventor： Kei Kikuiri ,  Nobuhiko Naka ,  Tomoyuki Ohya

IPC: G10L19/00

CPC classification number: G10L19/032 ,  G10L19/22 ,  H04N19/124 ,  H04N19/154 ,  H04N19/176 ,  H04N19/60 ,  H04N19/90

Abstract: The present invention intends to render quantization noise virtually imperceptible for a user and to prevent reduction in frequency resolution and reduction in encoding efficiency.A signal encoding apparatus includes: a quantization unit for quantizing an input signal based on a plurality of quantization methods; a dequantization unit for obtaining decoded signals by performing the dequantizing process; an error signal calculation unit for calculating a plurality of error signals between the decoded signals and the input signal; a weighting calculation unit for calculating, for each subblock, a weight related to degree concerning whether or not quantization noise corresponding to error signal is virtually imperceptible for a user; a quantization method selection unit for selecting a given quantization method from among the plurality of quantization methods, when a plurality of weighted error signals, obtained by assigning a weight of each subblock to an error signal of the subblock, are generated, based on the of weighted error signals; and an output unit for outputting the input signal quantized based on the given quantization method as an output signal.

Abstract translation: 本发明意图使得量化噪声对于用户实际上是不可察觉的，并且防止频率分辨率的降低和编码效率的降低。 信号编码装置包括：量化单元，用于基于多个量化方法量化输入信号; 去量化单元，用于通过执行去量化处理来获得解码信号; 误差信号计算单元，用于计算解码信号与输入信号之间的多个误差信号; 一个加权计算单元，用于针对每个子块计算与用户对于误差信号对应的量化噪声是否几乎不可察觉的程度相关的权重; 一种用于从多个量化方法中选择给定的量化方法的量化方法选择单元，当通过将每个子块的权重分配给子块的误差信号而获得的多个加权误差信号基于 加权误差信号; 以及输出单元，用于输出基于给定量化方法量化的输入信号作为输出信号。

公开(公告)号：US07406410B2

公开(公告)日：2008-07-29

申请号：US10359638

申请日：2003-02-07

Applicant: Kei Kikuiri ,  Nobuhiko Naka ,  Tomoyuki Ohya

Inventor： Kei Kikuiri ,  Nobuhiko Naka ,  Tomoyuki Ohya

IPC: G10L19/12

CPC classification number: G10L19/24 ,  G10L19/025

Abstract: A decoding apparatus is provided. The decoding apparatus has a first decoding part for decoding a code word obtained by encoding an input signal using a Code-Excited Linear Prediction encoding method. A second decoding part decodes a code word obtained by encoding a signal with an encoding method other than the Code-Excited Linear Prediction encoding method. A rising-transition detection and notification part has a detection part that detects the existence of a rising-transition of amplitude of the input signal based on time variation of a gain of excitation vectors obtained by the first decoding part, and a notification part that notifies the second decoding part that the rising-transition of the amplitude exists.

Abstract translation: 提供了解码装置。 解码装置具有第一解码部分，用于对通过使用码激励线性预测编码方法对输入信号进行编码而获得的码字进行解码。 第二解码部分通过使用除码激励线性预测编码方法之外的编码方法对信号进行编码而获得的码字进行解码。 上升转换检测和通知部分具有检测部分，其基于由第一解码部分获得的激励矢量的增益的时间变化来检测输入信号的振幅的上升转变的存在;以及通知部 第二解码部分存在振幅的上升转变。

公开(公告)号：US07363231B2

公开(公告)日：2008-04-22

申请号：US10646752

申请日：2003-08-25

Applicant: Kei Kikuiri ,  Nobuhiko Naka ,  Tomoyuki Ohya

Inventor： Kei Kikuiri ,  Nobuhiko Naka ,  Tomoyuki Ohya

IPC: G10L19/00 ,  G10L19/02

CPC classification number: G10L19/022 ,  G10L19/24

Abstract: A coding device capable of improving the coding efficiency and a decoding device for decoding a code sequence generated by the coding device are provided. In the coding device, for each of the possible block combinations obtained when dividing a frame, a coding unit encodes each block in the frame block by block at different bit rates, and at the same time, the coding unit decodes the resultant code sequences related to the frame. A calculation unit calculates the error powers of the decoded signals and the input signal. A determination unit selects a code sequence that makes the average bit rate in coding the frame not higher than a specified value and the corresponding error power a minimum. This selected code sequence is output.

Abstract translation: 提供能够提高编码效率的编码装置和用于对由编码装置生成的编码序列进行解码的解码装置。 在编码装置中，对于在划分帧时获得的每个可能的块组合，编码单元以不同的比特率对块中的每个块进行编码，并且同时，编码单元对相关的编码序列进行解码 到框架。 计算单元计算解码信号和输入信号的误差功率。 确定单元选择使帧的编码中的平均比特率不高于指定值并且相应的误差功率最小的编码序列。 输出该选择的代码序列。

公开(公告)号：US06522665B1

公开(公告)日：2003-02-18

申请号：US09269919

申请日：1999-04-01

Applicant: Takashi Suzuki ,  Toshio Miki ,  Toshiro Kawahara ,  Nobuhiko Naka

Inventor： Takashi Suzuki ,  Toshio Miki ,  Toshiro Kawahara ,  Nobuhiko Naka

IPC: H04J322

CPC classification number: H04L1/008 ,  G11B20/12 ,  G11B20/1217 ,  G11B20/1833 ,  G11B27/10 ,  G11B27/3027 ,  H03M13/09 ,  H04L1/0072 ,  H04L1/08

Abstract: The probability of frame destruction is lowered while suppressing the redundancy of the transmission data. On the transmitting side, a predetermined unique word is contained in a frame n for storing the n-th data, and header information n, frame length information and header information n−1 of the frame n−1 one frame before the frame n are subjected to error-correcting coding, contained in the frame n, and transmitted. On the receiving side, the header of the frame n is received. When the frame length information is transmitted with error, the timing is specified by detecting the unique word and header information in the next frame n+1. When the header of the frame n is not successfully decoded, the information data of the frame n is decoded by using the header information n inserted into a predetermined position of the frame n+1.

Abstract translation: 在抑制传输数据的冗余性的同时降低帧破坏的概率。 在发送侧，在用于存储第n个数据的帧n中包含预定的唯一字，并且帧n之前一帧的帧n-1的帧头信息n，帧长度信息和帧头信息n-1 经过包含在帧n中的纠错编码，并发送。 在接收侧，接收帧n的报头。 当错误地发送帧长度信息时，通过检测下一帧n + 1中的唯一字和标题信息来指定定时。 当帧n的报头未成功解码时，通过使用插入到帧n + 1的预定位置的标题信息n来解码帧n的信息数据。

公开(公告)号：US20120010879A1

公开(公告)日：2012-01-12

申请号：US13243015

申请日：2011-09-23

Applicant: Kosuke Tsujino ,  Kei Kikuiri ,  Nobuhiko Naka

Inventor： Kosuke Tsujino ,  Kei Kikuiri ,  Nobuhiko Naka

IPC: G10L19/04

CPC classification number: G10L19/03 ,  G10L19/0208 ,  G10L19/0212 ,  G10L19/06 ,  G10L19/167 ,  G10L19/24 ,  G10L19/26 ,  G10L21/038 ,  G10L21/04

Abstract: A linear prediction coefficient of a signal represented in a frequency domain is obtained by performing linear prediction analysis in a frequency direction by using a covariance method or an autocorrelation method. After the filter strength of the obtained linear prediction coefficient is adjusted, filtering may be performed in the frequency direction on the signal by using the adjusted coefficient, whereby the temporal envelope of the signal is transformed. This reduces the occurrence of pre-echo and post-echo and improves the subjective quality of the decoded signal, without significantly increasing the bit rate in a band extension technique in the frequency domain represented by SBR.

Abstract translation: 通过使用协方差方法或自相关方法通过在频率方向上进行线性预测分析来获得在频域中表示的信号的线性预测系数。 在调整所获得的线性预测系数的滤波强度之后，可以通过使用调整后的系数在频率方向对信号进行滤波，从而变换信号的时间包络。 这减少了前回波和后回波的发生，并提高了解码信号的主观质量，而不会显着增加由SBR表示的频域中的频带扩展技术中的比特率。

公开(公告)号：US07831209B2

公开(公告)日：2010-11-09

申请号：US11372056

申请日：2006-03-10

Applicant: Yasuyo Yasuda ,  Nobuhiko Naka ,  Tomoyuki Ohya ,  Toshio Miki

Inventor： Yasuyo Yasuda ,  Nobuhiko Naka ,  Tomoyuki Ohya ,  Toshio Miki

IPC: H04B1/38

CPC classification number: H04S1/005 ,  H04R27/00 ,  H04S7/303 ,  H04S2400/01 ,  H04S2400/11 ,  H04S2420/01

Abstract: To provide a data transmitter-receiver bidirectional transmitting system, and data transmitting-receiving method capable of reducing influences of a transmission delay when performing bidirectional communication with an other-communication-party apparatus in an environment in which various states are dynamically changed depending on time.A data obtaining portion 11 of a terminal 10 obtains positional directional information D(Tn) and a time information obtaining portion 12 obtains time information Tn. The terminal 10 stores D(Tn) and Tn in a memory portion 13 and a transmitting portion 15 transmits D(Tn) and Tn to a server 20. A data generating portion 21 of the server 20 generates stereophonic data S(Tn) by using D(Tn), a time information copying portion 22 copies Tn, a transmitting portion 24 transmits S(Tn) and Tn to the terminal 10. A data obtaining portion 11 of the terminal 10 obtains positional directional information D(Tm), the time information obtaining portion 12 obtains time information Tm, and a correcting portion 14 corrects the difference between D(Tm) and D(Tn) to generate S′(m).

Abstract translation: 提供一种数据收发双向发送系统和数据发送接收方法，能够在各种状态根据时间动态变化的环境中与其他通信方装置进行双向通信时减少传输延迟的影响 。 终端10的数据获取部分11获得位置方向信息D（Tn），时间信息获取部分12获得时间信息Tn。 终端10将D（Tn）和Tn存储在存储器部分13中，并且发送部分15将D（Tn）和Tn发送到服务器20.服务器20的数据产生部分21通过使用产生立体声数据S（Tn） D（Tn），时间信息复制部分22复制Tn，发送部分24向终端10发送S（Tn）和Tn。终端10的数据获取部分11获得位置方向信息D（Tm） 信息获取部分12获得时间信息Tm，并且校正部分14校正D（Tm）和D（Tn）之间的差异以产生S'（m）。

公开(公告)号：US06985501B2

公开(公告)日：2006-01-10

申请号：US09827639

申请日：2001-04-06

Applicant: Takashi Suzuki ,  Toshiro Kawahara ,  Masashi Morioka ,  Nobuhiko Naka

Inventor： Takashi Suzuki ,  Toshiro Kawahara ,  Masashi Morioka ,  Nobuhiko Naka

IPC: H04J3/06 ,  H04L12/26

CPC classification number: H04L65/80 ,  H04L12/6418 ,  H04L29/06027 ,  H04L47/10 ,  H04L47/2416 ,  H04L47/28 ,  H04L2012/6472 ,  H04L2012/6489

Abstract: A delay unit 103 adds holding time that has been set by a holding time setting unit 104 to a received data. The holding time is computed based on delay time of received data and the minimum delay time of data received up to a certain point for the purpose of reducing a total delay time. The delay time is estimated in a delay time estimating unit 106 from the difference between a reception time of a packet counted based on an internal clock generator 107 and a time designated by a time stamp in the received packet.