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    • 2. 发明授权
    • Sound signal processor and sound signal processing methods
    • 声音信号处理器和声音信号处理方法
    • US08873766B2
    • 2014-10-28
    • US13345220
    • 2012-01-06
    • Toshifumi YamamotoTadashi Amada
    • Toshifumi YamamotoTadashi Amada
    • H04R29/00H03G9/00
    • H04R29/001
    • According to one embodiment, a sound signal processor includes: a connector; an input module; and a generator. The connector is connectable with an earphone. The input module receives and processes a plurality of sound signals corresponding to sound of a plurality of times output from the earphone, respectively. The generator generates, by using first data indicating a frequency characteristic of a first sound signal among the received and processed sound signals and second data indicating a frequency characteristic of a second sound signal among the received and processed sound signals, correction data correcting a frequency characteristic of the earphone to be a target frequency characteristic set as a target. The first data is used for a first frequency band lower than or equal to a reference. The second data is used for a second frequency band higher than the reference.
    • 根据一个实施例,声音信号处理器包括:连接器; 一个输入模块; 和发电机。 连接器可与耳机连接。 输入模块分别接收和处理对应于从耳机输出的多次声音的多个声音信号。 发生器通过使用指示接收和处理的声音信号中的第一声音信号的频率特性的第一数据和指示接收和处理的声音信号中的第二声音信号的频率特性的第二数据,校正数据校正频率特性 的耳机作为目标频率特性。 第一数据用于低于或等于参考的第一频带。 第二数据用于高于参考的第二频带。
    • 4. 发明申请
    • SOUND SIGNAL PROCESSOR AND SOUND SIGNAL PROCESSING METHODS
    • 声信号处理器和声信号处理方法
    • US20120275616A1
    • 2012-11-01
    • US13345220
    • 2012-01-06
    • Toshifumi YamamotoTadashi Amada
    • Toshifumi YamamotoTadashi Amada
    • H04R1/10
    • H04R29/001
    • According to one embodiment, a sound signal processor includes: a connector; an input module; and a generator. The connector is connectable with an earphone. The input module receives and processes a plurality of sound signals corresponding to sound of a plurality of times output from the earphone, respectively. The generator generates, by using first data indicating a frequency characteristic of a first sound signal among the received and processed sound signals and second data indicating a frequency characteristic of a second sound signal among the received and processed sound signals, correction data correcting a frequency characteristic of the earphone to be a target frequency characteristic set as a target. The first data is used for a first frequency band lower than or equal to a reference. The second data is used for a second frequency band higher than the reference.
    • 根据一个实施例,声音信号处理器包括:连接器; 一个输入模块; 和发电机。 连接器可与耳机连接。 输入模块分别接收和处理对应于从耳机输出的多次声音的多个声音信号。 发生器通过使用指示接收和处理的声音信号中的第一声音信号的频率特性的第一数据和指示接收和处理的声音信号中的第二声音信号的频率特性的第二数据,校正数据校正频率特性 的耳机作为目标频率特性。 第一数据用于低于或等于参考的第一频带。 第二数据用于高于参考的第二频带。
    • 6. 发明授权
    • Signal processing method, apparatus and program
    • 信号处理方法,装置和程序
    • US08630850B2
    • 2014-01-14
    • US13240353
    • 2011-09-22
    • Kaoru SuzukiTadashi Amada
    • Kaoru SuzukiTadashi Amada
    • G10L21/00
    • H04B3/23H04M9/082
    • In one embodiment, a signal processing method is disclosed. The method can perform filter processing of convoluting a tap coefficient in a first signal sequence to generate a second signal sequence. The method can subtract the second signal sequence from a third signal sequence to generate a fourth signal sequence. The third signal sequence includes an echo signal of the first signal sequence. The method can correct the tap coefficient in accordance with an amount of correction determined using a function. The function includes at least one of a first region and a second region, and has values limited. The first region is included in a negative value region of the fourth signal sequence. The second region is included in a positive value region of the fourth signal sequence.
    • 在一个实施例中,公开了一种信号处理方法。 该方法可以执行在第一信号序列中卷积抽头系数的滤波处理以产生第二信号序列。 该方法可以从第三信号序列中减去第二信号序列以产生第四信号序列。 第三信号序列包括第一信号序列的回波信号。 该方法可以根据使用功能确定的校正量来校正抽头系数。 该功能包括第一区域和第二区域中的至少一个,并且具有限制的值。 第一区域被包括在第四信号序列的负值区域中。 第二区域被包括在第四信号序列的正值区域中。
    • 8. 发明申请
    • PICKUP SIGNAL PROCESSING APPARATUS, METHOD, AND PROGRAM PRODUCT
    • PICKUP信号处理设备,方法和程序产品
    • US20110313763A1
    • 2011-12-22
    • US13219844
    • 2011-08-29
    • Tadashi Amada
    • Tadashi Amada
    • G10L15/20H04R3/00
    • H04R3/005G10L25/78G10L2021/02165H04R2430/20
    • According to one embodiment, a pickup signal processing apparatus includes microphones, a sound determining unit, a signal level calculating unit, a setting unit, and a calculating unit. The sound determining unit determines whether pickup signals picked up by the microphones are signals from a neighboring sound source or a background noise signal. The signal level calculating unit calculates the signal levels for the microphones. The setting unit sets a gain value of at least one microphone and reduces a difference between the signal levels for the microphones on the basis of the signal levels for the microphones, when determined that the pickup signal is the background noise signal. The calculating unit multiplies the pickup signal of the at least one microphone by the gain value set by the setting unit.
    • 根据一个实施例,拾取信号处理装置包括麦克风,声音确定单元,信号电平计算单元,设置单元和计算单元。 声音确定单元确定由麦克风拾取的拾取信号是否是来自相邻声源的信号或背景噪声信号。 信号电平计算单元计算麦克风的信号电平。 当确定拾取信号是背景噪声信号时,设置单元设置至少一个麦克风的增益值,并且基于麦克风的信号电平来减小麦克风的信号电平之间的差异。 计算单元将至少一个麦克风的拾取信号乘以由设置单元设置的增益值。
    • 10. 发明授权
    • Speech coding/decoding method and apparatus
    • 语音编码/解码方法及装置
    • US06611797B1
    • 2003-08-26
    • US09488748
    • 2000-01-21
    • Tadashi AmadaKatsumi Tsuchiya
    • Tadashi AmadaKatsumi Tsuchiya
    • G10L1910
    • G10L19/12
    • An input speech signal to an input terminal is supplied to a speech synthesizer section through a speech analyzer section and frequency parameter quantizer section to form a synthesis filter, and the input speech signal is expressed by quantized LPC coefficients representing the characteristics of the synthesis filter and an excitation signal for exciting the synthesis filter. In this case, in a pulse excitation section, a pulse position selector selects pulse position candidates from the integer pulse positions and non-integer pulse positions stored in a pulse position codebook, and an integer position pulse generator and non-integer position pulse generator respectively generate integer position pulses set at sampling points of the excitation signal and non-integer position pulses set at positions located between sampling points. These pulses are synthesized into a pulse train serving as a source of an excitation signal.
    • 通过语音分析器部分和频率参数量化器部分向输入终端输入的语音信号提供给语音合成器部分以形成合成滤波器,并且输入语音信号由表示合成滤波器的特性的量化LPC系数表示, 用于激发合成滤波器的激励信号。 在这种情况下,在脉冲激励部中,脉冲位置选择器分别从存储在脉冲位置码本中的整数脉冲位置和非整数脉冲位置选择脉冲位置候补,分别选择整数位置脉冲发生器和非整数位置脉冲发生器 产生在激励信号的采样点设置的整数位置脉冲和设置在采样点之间的位置处的非整数位置脉冲。 这些脉冲被合成为用作激励信号的源的脉冲串。