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    • 1. 发明授权
    • Speech decoding method and apparatus which generates an excitation signal and a synthesis filter
    • 产生激励信号和合成滤波器的语音解码方法和装置
    • US08249866B2
    • 2012-08-21
    • US12751191
    • 2010-03-31
    • Kimio Miseki
    • Kimio Miseki
    • G10L19/08G10L19/12
    • G10L19/18
    • A speech decoding method which generates an excitation signal and a synthesis filter from coded data and which obtains a speech signal based on the excitation signal and the synthesis filter. The method includes acquiring identification information used for determining whether the speech signal to be decoded is a narrowband signal or a wideband signal; and modifying the excitation signal based on the identification information by controlling strength or presence of emphasis of pitch periodicity with respect to the excitation signal generated from the coded data, so as to generate the speech signal by use of the modified excitation signal and the synthesis filter.
    • 一种语音解码方法,其从编码数据生成激励信号和合成滤波器,并且基于激励信号和合成滤波器获得语音信号。 该方法包括获取用于确定要解码的语音信号是窄带信号还是宽带信号的识别信息; 以及通过相对于从编码数据产生的激励信号控制音调周期的强度的强度或存在,基于识别信息修改激励信号,以便通过使用修改的激励信号和合成滤波器来产生语音信号 。
    • 2. 发明申请
    • ACOUSTIC SIGNAL CORRECTOR AND ACOUSTIC SIGNAL CORRECTING METHOD
    • 声学信号校正和声学信号校正方法
    • US20110206220A1
    • 2011-08-25
    • US12949437
    • 2010-11-18
    • Norikatsu ChibaKimio Miseki
    • Norikatsu ChibaKimio Miseki
    • H03G3/00
    • H04R1/1091G10L21/02G10L21/0232H03G5/005H03G9/005H04R1/1016H04R25/353H04R25/356H04R2460/15
    • According to one embodiment, an acoustic signal corrector includes: an output module; a selection receiver; and a holder. The output module is configured to output a plurality of acoustic signals. Amplitude values of frequencies within a frequency band of each of the acoustic signals are emphasized as emphasized amplitude values. A plurality of amplitude values among the emphasized amplitude values are corrected as corrected amplitude values at some of frequencies within the frequency band. Resonance is possibly induced in the frequency band by sealing an ear canal. The selection receiver is configured to receive a selection of one of the acoustic signals output by the output module. The holder is configured to hold, as a configuration for sound quality correction, a configuration corresponding to the correction of the one of the acoustic signals at the some of the frequencies.
    • 根据一个实施例,声信号校正器包括:输出模块; 选择接收机; 和持有人。 输出模块被配置为输出多个声信号。 每个声信号的频带内的频率的振幅值被强调为强调的振幅值。 强调幅度值中的多个振幅值被校正为在频带内的一些频率处的校正振幅值。 通过密封耳道可能在频带中引起共振。 选择接收器被配置为接收由输出模块输出的声信号之一的选择。 保持器被配置为作为用于声音质量校正的配置来保持与在一些频率处的声信号之一的校正相对应的配置。
    • 3. 发明授权
    • Content-reproducing apparatus
    • 内容再现装置
    • US07698296B2
    • 2010-04-13
    • US11699814
    • 2007-01-30
    • Chikashi SugiuraTakehiko IsakaKimio Miseki
    • Chikashi SugiuraTakehiko IsakaKimio Miseki
    • G06F7/00
    • G06F17/30029G06F17/30053Y10S707/99943
    • A content reproducing apparatus includes a display unit configured to display a play list and candidate contents able to be added to the play list, a selection unit configured to select, from the candidate contents, an undesired content which a user does not want to add to the play list, a calculation unit configured to calculate a first retrieval statistical quantity based on first characteristic quantity of the undesired content or a second retrieval statistical quantity based on second characteristic quantity of a desired content which the user wants to add to the play list, and a retrieve unit configured to retrieve the candidate contents to prepare the play list, in accordance with similarity which has been calculated by using the first or second retrieval statistical quantity and which shows to which a given content having third characteristic quantity is similar, the desired content or the undesired content.
    • 内容再现装置包括:显示单元,被配置为显示播放列表和能够添加到播放列表的候选内容;选择单元,被配置为从候选内容中选择用户不想添加的不需要的内容 播放列表,计算单元,被配置为基于用户想要添加到播放列表的期望内容的第二特征量,基于不需要内容的第一特征量或第二检索统计量来计算第一检索统计量, 以及检索单元,其被配置为根据已经通过使用第一或第二检索统计量计算并且示出具有第三特征量的给定内容相似的相似度来检索候选内容以准备播放列表,期望的 内容或不需要的内容。
    • 5. 发明申请
    • Information Processing Apparatus and Program
    • 信息处理装置与程序
    • US20080247557A1
    • 2008-10-09
    • US12045457
    • 2008-03-10
    • Takashi SudoKimio MisekiYuji Kawashima
    • Takashi SudoKimio MisekiYuji Kawashima
    • H04B3/20
    • H04M9/082
    • According to one embodiment, a signal processing apparatus includes a speaker configured to output the received input signal on which a delay detection signal which has a frequency component of an inaudible frequency on a received input signal is superposed to an acoustic space, an extracting section configured to extract the delay detection signal from the sending input signal outputted from microphone configured to collect sound in the acoustic space a calculating section configured to calculate a delay time between the received input signal and an acoustic echo component contained in the sending input signal, a delay section configured to delay the received input signal by a time corresponding to the delay time and generate a delayed received input signal, and an echo suppression processing section configured to suppress the acoustic echo component contained in the sending input signal by use of the delayed received input signal.
    • 根据一个实施例,一种信号处理装置包括:扬声器,被配置为输出接收到的输入信号,在所述接收输入信号上将具有接收到的输入信号的不可听频率的频率分量的延迟检测信号叠加到声学空间;提取部分配置 从被配置为在声学空间中收集声音的麦克风输出的发送输入信号中提取延迟检测信号,计算部分,被配置为计算接收的输入信号和包含在发送输入信号中的声学回声分量之间的延迟时间,延迟 被配置为将接收的输入信号延迟与延迟时间相对应的时间,并产生延迟的接收输入信号;以及回波抑制处理部,被配置为通过使用延迟的接收输入来抑制包含在发送输入信号中的声学回声分量 信号。
    • 7. 发明授权
    • Background noise/speech classification method
    • 背景噪音/语音分类方法
    • US06202046B1
    • 2001-03-13
    • US09012792
    • 1998-01-23
    • Masahiro OshikiriKimio MisekiMasami Akamine
    • Masahiro OshikiriKimio MisekiMasami Akamine
    • G10L1106
    • G10L25/93G10L19/09G10L25/78
    • In a background noise/speech classification method, whether a digital input signal input through an input terminal is background noise or speech is decided by a background noise/speech decision section on the basis of calculated frame power and a calculated LSP coefficient which are obtained by supplying the input signal to a feature amount calculation section and estimated frame power and an estimated LSP coefficient obtained by an estimated feature amount update section. Thereafter, the estimated feature amount update section updates the estimated frame power and the estimated LSP coefficient by using the frame power and the LSP coefficient obtained by the feature amount calculation section to prepare for the next frame.
    • 在背景噪声/语音分类方法中,基于计算出的帧功率和计算出的LSP系数,通过背景噪声/语音判定部分确定通过输入端子输入的数字输入信号是背景噪声还是语音是由 将输入信号提供给特征量计算部分,并且估计帧功率和由估计特征量更新部分获得的估计LSP系数。 此后,估计特征量更新部分通过使用由特征量计算部分获得的帧功率和LSP系数来准备下一帧来更新估计的帧功率和估计的LSP系数。
    • 8. 再颁专利
    • Speech coding and decoding apparatus
    • 语音编解码装置
    • USRE36721E
    • 2000-05-30
    • US561751
    • 1995-11-22
    • Masami AkamineKimio Miseki
    • Masami AkamineKimio Miseki
    • G10L9/00
    • A speech signal is input to an excitation signal generating section, a prediction filter and a prediction parameter calculator. The prediction parameter calculator calculates a predetermined number of prediction parameters (LPC parameter or reflection coefficient) by an autocorrelation method or covariance method, and supplies the acquired prediction parameters to a prediction parameter coder. The codes of the prediction parameters are sent to a decoder and a multiplexer. The decoder sends decoded values of the codes of the prediction parameters to the prediction filter and the excitation signal generating section. The prediction filter calculates a prediction residual signal, which is the difference between the input speech signal and the decoded prediction parameter, and sends it to the excitation signal generating section. The excitation signal generating section calculates the pulse interval and amplitude for each of a predetermined number of subframes based on the input speech signal, the prediction residual signal and the quantized value of the prediction parameter, and sends them to the multiplexer. The multiplexer combines these codes and the codes of the prediction parameters, and send the results as an output signal of a coding apparatus to a transmission path or the like.
    • 语音信号被输入到激励信号产生部分,预测滤波器和预测参数计算器。 预测参数计算器通过自相关方法或协方差方法计算预定数量的预测参数(LPC参数或反射系数),并将所获取的预测参数提供给预测参数编码器。 预测参数的代码被发送到解码器和多路复用器。 解码器将预测参数的代码的解码值发送到预测滤波器和激励信号生成部。 预测滤波器计算作为输入语音信号和解码预测参数之间的差的预测残差信号,并将其发送到激励信号生成部。 激励信号生成部基于输入的语音信号,预测残差信号和预测参数的量化值,计算预定数量的子帧中的每一个的脉冲间隔和幅度,并将其发送到多路复用器。 多路复用器组合这些代码和预测参数的代码,并将结果作为编码装置的输出信号发送到传输路径等。