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    • 11. 发明授权
    • Speech encoding apparatus utilizing stored code data
    • 利用存储码数据的语音编码装置
    • US5671327A
    • 1997-09-23
    • US7710
    • 1993-01-22
    • Masami AkamineMasahiro OshikiriKimio Miseki
    • Masami AkamineMasahiro OshikiriKimio Miseki
    • G10L19/00G10L19/12G10L3/02G10L9/00
    • G10L19/12
    • A learning-type speech encoding apparatus comprises an adaptive code book storing driving signal vectors, a minimum distortion searching circuit for searching the adaptive code book for an optimum driving signal vector on the basis of the input speech signal, a synthesizing filter for synthesizing a speech signal using the optimum driving signal vector retrieved, a buffer for storing the optimum driving signal vector retrieved, a training vector creating section for producing a training vector by segmenting the stored driving signal vector in units of a specified length, and a learning section for learning by constantly updating the driving signal vectors in the code book on the basis of the training vector.
    • 一种学习型语音编码装置,包括存储驱动信号矢量的自适应码本,用于基于输入的语音信号搜索自适应码本以获得最佳驱动信号向量的最小失真搜索电路,用于合成语音的合成滤波器 使用所检索的最佳驱动信号矢量的信号,用于存储所检索的最佳驱动信号矢量的缓冲器,用于通过以指定长度为单位分割存储的驾驶信号向量来产生训练矢量的训练矢量创建部分,以及用于学习的学习部分 通过基于训练矢量不断更新码本中的驾驶信号矢量。
    • 12. 发明授权
    • Variable rate encoding and communicating apparatus
    • 可变速率编码和通信装置
    • US5150387A
    • 1992-09-22
    • US630911
    • 1990-12-20
    • Hidetaka YoshikawaKimio MisekiMasami Akamine
    • Hidetaka YoshikawaKimio MisekiMasami Akamine
    • H04B1/66
    • H04B1/667G10L19/24
    • In a transmitter in the present invention, an input signal is input to a QMF bank 102 where the input signal is divided to a plurality of frequency bands to form corresponding band signals. A distributed bit calculating unit 109 calculates respective bit rates with which the corresponding band signals are encoded on the respective power values of the band signals. Quantizers 104-1, 104-2, . . . , 104-n encode the respective band signals at the corresponding bit rates and input the resulting corresponding band codes to a multiplexer unit 111 which incorporates the respective band codes into a cell as an information unit and sends the cell. In a receiver, a cell is decomposed to obtain the respective band codes, which are then dequantized to form the corresponding band signals. These band signals are synthesized to form a signal for the entire band, and the signal for the entire band is output as a decoded signal.
    • 在本发明的发射机中,将输入信号输入到QMF组102,其中输入信号被划分成多个频带以形成相应的频带信号。 分布位计算单元109计算相应频带信号对频带信号的各个功率值进行编码的各个比特率。 量子化器104-1,104-2, 。 。 ,104-n以对应的比特率对各个频带信号进行编码,并将所得到的相应频带码输入到将各频带码合并到一个小区中作为信息单元的多路复用单元111,并发送该小区。 在接收机中,单元被分解以获得相应的频带码,然后将它们去量化以形成相应的频带信号。 这些频带信号被合成以形成整个频带的信号,并且将整个频带的信号作为解码信号输出。
    • 13. 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。
    • 14. 发明授权
    • Speech coding and decoding apparatus
    • 语音编解码装置
    • US5265167A
    • 1993-11-23
    • US13551
    • 1992-11-19
    • Masami AkamineKimio Miseki
    • Masami AkamineKimio Miseki
    • G10L19/04G10L19/10G10L9/00
    • G10L19/113
    • 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。
    • 16. 发明授权
    • Method and system for speech encoding involving analyzing search range for current period according to length of preceding pitch period
    • 用于语音编码的方法和系统,涉及根据前一音调周期的长度分析当前周期的搜索范围
    • US06470310B1
    • 2002-10-22
    • US09407060
    • 1999-09-28
    • Masahiro OshikiriKimio Miseki
    • Masahiro OshikiriKimio Miseki
    • G10L1104
    • G10L25/90G10L19/08
    • Processing for producing encoded output representing information about a pitch period of an input speech signal is performed. The pitch period of a previously entered speech signal is stored in a buffer. A search range-determining portion determines a range in which a current pitch period is analyzed, according to the pitch period of the previously entered speech signal. A presently entered speech signal is applied from a speech input terminal. A pitch analysis portion makes a pitch analysis of candidates for the pitch period contained in the determined search range. Information about the pitch period is delivered from an output terminal and stored in the buffer for subsequent processing. The pitch period of the speech signal can be calculated with a small amount of calculation and represented with a small amount of information.
    • 执行用于产生表示关于输入语音信号的音调周期的信息的编码输出的处理。 先前输入的语音信号的音调周期被存储在缓冲器中。 搜索范围确定部分根据先前输入的语音信号的音调周期来确定当前音调周期被分析的范围。 从语音输入端子应用当前输入的语音信号。 音调分析部分对包含在确定的搜索范围内的音调周期的候选进行音调分析。 关于音调周期的信息从输出端传送并存储在缓冲器中用于后续处理。 可以用少量的计算来计算语音信号的音调周期,并用少量的信息表示。
    • 18. 发明授权
    • Speech encoding/decoding method using reduced subframe pulse positions having density related to pitch
    • 使用具有与音调相关的密度的减少的子帧脉冲位置的语音编码/解码方法
    • US06385576B2
    • 2002-05-07
    • US09220062
    • 1998-12-23
    • Tadashi AmadaKimio Miseki
    • Tadashi AmadaKimio Miseki
    • G10L1908
    • G10L19/10
    • A speech encoding method in which information representing characteristics of a synthesis filter is generated based on an input speech signal in units of one frame. A pitch vector is generated from an adaptive codebook containing past excitation signals, and a first number of reduced pulse position candidates are generated by selecting a first number of pulse positions from a number of possible pulse positions in each of sub-frames obtained by dividing the frame, where a density of the reduced pulse position candidates is high where the pitch vector has a large power and decreases in accordance with a decrease in the power. A second number of pulse positions is selected from the reduced pulse position candidates to generate a pulse train having a plurality of pulses located at pulse positions corresponding to a second number of pulse positions under the criterion of minimizing an error between the input speech signal and a synthesis signal which is an output of the synthesis filter whose input is an excitation signal generated by adding the pitch vector and the pulse train.
    • 一种语音编码方法,其中基于一帧的单位的输入语音信号来生成表示合成滤波器的特性的信息。 从包含过去的激励信号的自适应码本生成音调矢量,并且通过从通过划分所得到的每个子帧中的每个子帧中的多个可能的脉冲位置中选择第一数量的脉冲位置来生成第一数量的缩小脉冲位置候选 帧,其中缩小脉冲位置候选的密度高,其中音调矢量具有大的功率,并且根据功率的降低而减小。 从缩小的脉冲位置候选中选择第二数量的脉冲位置,以产生脉冲序列,该脉冲串具有位于与第二数量的脉冲位置相对应的脉冲位置的多个脉冲,该标准在最小化输入语音信号与a 合成信号是合成滤波器的输出,其输入是通过将音调矢量和脉冲串相加而产生的激励信号。