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    • 1. 发明授权
    • Adaptive differential pulse code modulation encoding apparatus and decoding apparatus
    • 自适应差分脉码调制编码装置和解码装置
    • US08482439B2
    • 2013-07-09
    • US13142010
    • 2009-12-25
    • Yasushi SatoAtsuko Ryu
    • Yasushi SatoAtsuko Ryu
    • H03M7/32
    • G10L19/04H03M7/3046
    • A signal corresponding to a short-period change and a signal corresponding to a long-period change of a sound signal are detected, and optimal quantization is performed based on the combination of the two signals. In an ADPCM encoding apparatus (100), a differential value dn between a 16-bit input signal Xn and a decoded signal Yn-1 of one sample ago is calculated by a subtractor (102). Thereafter, the 16-bit differential value dn is adaptively quantized by an adaptive quantizing section (103), so as to be converted to a (1 to 8)-bit length-variable ADPCM value Dn. Thereafter, the ADPCM value Dn is compression-encoded by a compression-encoding section (108) to generate a signal D′n, and the signal D′n is framed by a framing section (130) and outputted. Further, in an ADPCM decoding apparatus, a framed input signal is subjected to a reverse of the aforesaid process so as to be decoded.
    • 检测对应于短周期变化的信号和对应于声音信号的长周期变化的信号,并且基于两个信号的组合来执行最佳量化。 在ADPCM编码装置(100)中,通过减法器(102)计算16位输入信号Xn与一个采样前的解码信号Yn-1之间的微分值dn。 此后,通过自适应量化部分(103)自适应地量化16位微分值dn,以便将其转换为(1至8)位长度可变ADPCM值Dn。 此后,ADPCM值Dn由压缩编码部(108)进行压缩编码,生成信号D'n,信号D'n由成帧部(130)构成并输出。 此外,在ADPCM解码装置中,对帧输入信号进行与上述处理相反的处理,以进行解码。
    • 2. 发明申请
    • ACTIVE MUFFLER
    • 主动式MUFFLER
    • US20110110527A1
    • 2011-05-12
    • US13001937
    • 2009-07-08
    • Yasushi SatoAtsuko Ryu
    • Yasushi SatoAtsuko Ryu
    • G10K11/16
    • G10K11/178G10K2210/12G10K2210/3212H04R3/02H04R3/04H04R2400/00
    • In an active muffler having improved response characteristics, a speaker section includes a diaphragm adapted to generate sound, a voice coil for driving the diaphragm, and a distance sensor to detect the movement of the diaphragm. A light generated by the LED is reflected by the diaphragm, the reflected light is detected by a phototransistor to thereby measure the distance to the diaphragm, so that the movement of the diaphragm is detected. Noise is detected by a microphone, and a signal having opposite phase to that of the noise is generated by an opposite-phase generating section. The difference between the opposite-phase signal and the signal of the distance to the speaker from the distance sensor is calculated and inputted to a PID control section. Such a difference indicates the delay of the speaker movement. Feedback control is performed in a direction in which the difference is canceled out.
    • 在具有改善的响应特性的主动消音器中,扬声器部分包括适于产生声音的隔膜,用于驱动隔膜的音圈和用于检测隔膜移动的距离传感器。 由LED产生的光被光阑反射,反射光被光电晶体管检测,从而测量与光阑的距离,从而检测光阑的移动。 由麦克风检测噪声,并且由相反相位产生部产生与噪声相反相位的信号。 计算相对相位信号与距距离传感器到扬声器距离的信号之间的差异,并输入到PID控制部。 这种差异表示扬声器运动的延迟。 在差异抵消的方向上进行反馈控制。
    • 3. 发明授权
    • Active muffler
    • 主动消声器
    • US08917879B2
    • 2014-12-23
    • US13001937
    • 2009-07-08
    • Yasushi SatoAtsuko Ryu
    • Yasushi SatoAtsuko Ryu
    • G10K11/16G10K11/178H04R3/02H04R3/04
    • G10K11/178G10K2210/12G10K2210/3212H04R3/02H04R3/04H04R2400/00
    • In an active muffler having improved response characteristics, a speaker section includes a diaphragm adapted to generate sound, a voice coil for driving the diaphragm, and a distance sensor to detect the movement of the diaphragm. A light generated by the LED is reflected by the diaphragm, the reflected light is detected by a phototransistor to thereby measure the distance to the diaphragm, so that the movement of the diaphragm is detected. Noise is detected by a microphone, and a signal having opposite phase to that of the noise is generated by an opposite-phase generating section. The difference between the opposite-phase signal and the signal of the distance to the speaker from the distance sensor is calculated and inputted to a PID control section. Such a difference indicates the delay of the speaker movement. Feedback control is performed in a direction in which the difference is canceled out.
    • 在具有改善的响应特性的主动消音器中,扬声器部分包括适于产生声音的隔膜,用于驱动隔膜的音圈和用于检测隔膜移动的距离传感器。 由LED产生的光被光阑反射,反射光被光电晶体管检测,从而测量与光阑的距离,从而检测光阑的移动。 由麦克风检测噪声,并且由相反相位产生部产生与噪声相反相位的信号。 计算相对相位信号与距距离传感器到扬声器距离的信号之间的差异,并输入到PID控制部。 这种差异表示扬声器运动的延迟。 在差异抵消的方向上进行反馈控制。
    • 4. 发明授权
    • High frequency signal interpolating apparatus
    • 高频信号插值装置
    • US08666732B2
    • 2014-03-04
    • US12311367
    • 2007-10-16
    • Yasushi SatoAtsuko Ryu
    • Yasushi SatoAtsuko Ryu
    • G10L19/02H03K5/24
    • G10L21/038
    • A high frequency signal interpolation apparatus provides, with a simple structure, a high-quality digital audio signal through interpolation of high frequency signals missing due to compression. The high frequency signal interpolation apparatus includes a peak value detection and holding circuit configured to detect a peak value of a digital audio signal provided to an input terminal by sampling the digital audio signal and generate a square wave signal by holding the detected peak value; a high-pass filter configured to extract a higher harmonic component from the generated square wave signal; and an adder configured to add the extracted higher harmonic component to the digital audio signal provided to the input terminal.
    • 高频信号插值装置通过内插由于压缩而丢失的高频信号,以简单的结构提供高质量的数字音频信号。 高频信号插值装置包括峰值检测和保持电路,其被配置为通过对数字音频信号进行采样来检测提供给输入端的数字音频信号的峰值,并通过保持检测到的峰值产生方波信号; 高通滤波器,被配置为从所产生的方波信号中提取高次谐波分量; 以及加法器,被配置为将提取的高次谐波分量添加到提供给输入端的数字音频信号。
    • 5. 发明授权
    • High-frequency signal interpolation apparatus and high-frequency signal interpolation method
    • 高频信号插值装置和高频信号插值方法
    • US08301281B2
    • 2012-10-30
    • US12448419
    • 2007-12-25
    • Yasushi SatoAtsuko Ryu
    • Yasushi SatoAtsuko Ryu
    • G06F17/00
    • G10L21/038
    • A favorable high-frequency signal is generated and practical high-frequency signal interpolation is implemented through simple processing. A digital audio signal reproduced by an instrument, which also carries out compression, is supplied as an original signal to an input terminal 1, and this original signal is then supplied to a digital sample and hold circuit 3 via a band-pass filter 2. The signal from the digital sample and hold circuit 3 is supplied to a ±1 multiplier 6, which then alternately inverts sign bits. The harmonic components of this signal in which the sign bits are inverted alternately are extracted by a high-pass filter (HPF) 7. Meanwhile, the original signal from the input terminal 1 is supplied to a delay circuit 8 equivalent to the processing time consumed by the aforementioned digital sample and hold circuit 3 and related circuits, forming an adjusted, delayed signal. The signals from the high-pass filter (HPF) 7 and the delay circuit 8 are then added by an adder 10, and the resulting added signal is then output to an output terminal 11.
    • 产生有利的高频信号,通过简单的处理实现了实用的高频信号插值。 由进行压缩的仪器再现的数字音频信号作为原始信号提供给输入端1,然后该原始信号经由带通滤波器2被提供给数字采样和保持电路3。 来自数字采样和保持电路3的信号被提供给±1乘法器6,然后交替地反转符号位。 通过高通滤波器(HPF)7提取符号位交替反转的该信号的谐波分量。同时,来自输入端1的原始信号被提供给等效于所消耗的处理时间的延迟电路8 通过上述数字采样保持电路3和相关电路,形成经调整的延迟信号。 来自高通滤波器(HPF)7和延迟电路8的信号然后由加法器10相加,然后将所得到的相加信号输出到输出端子11。
    • 6. 发明申请
    • High frequency signal interpolating method and high frequency signal interpolating
    • 高频信号内插方法和高频信号内插
    • US20100023333A1
    • 2010-01-28
    • US12311367
    • 2007-10-16
    • Yasushi SatoAtsuko Ryu
    • Yasushi SatoAtsuko Ryu
    • G10L21/00
    • G10L21/038
    • A quality high frequency signal is generated through simple processing, and practical high frequency signal interpolation is carried out. A digital audio signal reproduced by an apparatus, which carries out compression, is provided to an input terminal 1 as an original signal. This original signal is sent to a peak value detection and holding circuit 2, which detects and holds a peak value and then generates a square wave signal; wherein a higher harmonic component is included in the square wave signal.This higher harmonic component is extracted by a high pass filter (HPF) 3. On the other hand, the original signal from the input terminal 1 is sent to a delay circuit 4, which delays it for a time equivalent to the processing time of the peak value detection and holding circuit 2 described above. The resulting delayed, aligned signal is sent to a low pass filter (LPF) 5, which then generates a high frequency component-removed signal. The signals from the high pass filter (HPF) 3 and the low pass filter (LPF) 5 are then added by an adder 6, which then outputs the resulting signal to an output terminal 7.
    • 通过简单的处理产生高质量的高频信号,进行实际的高频信号插值。 由进行压缩的装置再生的数字音频信号作为原始信号提供给输入端1。 该原始信号被发送到峰值检测和保持电路2,其检测并保持峰值,然后产生方波信号; 其中在方波信号中包括高次谐波分量。 这种高次谐波分量由高通滤波器(HPF)3提取。另一方面,来自输入端子1的原始信号被发送到延迟电路4,延迟电路4等待时间等于处理时间 峰值检测和保持电路2。 所产生的延迟对齐信号被发送到低通滤波器(LPF)5,然后产生高频分量消除信号。 来自高通滤波器(HPF)3和低通滤波器(LPF)5的信号然后由加法器6相加,加法器6然后将所得到的信号输出到输出端子7。
    • 7. 发明授权
    • Acoustic characteristic control apparatus
    • 声学特性控制装置
    • US08242836B2
    • 2012-08-14
    • US13057896
    • 2009-08-03
    • Yasushi SatoAtsuko Ryu
    • Yasushi SatoAtsuko Ryu
    • H03K5/00
    • G10L25/48G10H1/0008G10H2210/061G10L25/09
    • An acoustic characteristic control apparatus supplies music signal, for example, to input terminal connected to a band-pass filter and a peaking filter. In a zero-cross detection circuit, a pulse signal corresponding to a period while a signal is positive is formed. A pulse-width measuring circuit output a signal corresponding to a pulse width. Next, the output of the pulse-width measuring circuit is inputted to one comparator and another comparator. The one comparator discriminates a time when the pulse width is equal to or larger than a first setting value, and the another comparator discriminates a time when the pulse width is equal to or smaller than a second setting value. The comparator is connected to the up terminal and the down terminal of an up/down counter. The output of the up/down counter is connected to the peaking filter through the subtractor, and acoustic characteristics of the peaking filter is controlled according to the count value of the up/down counter.
    • 声学特性控制装置例如将音乐信号提供给连接到带通滤波器和峰值滤波器的输入端子。 在零交叉检测电路中,形成与信号为正的周期相对应的脉冲信号。 脉冲宽度测量电路输出与脉冲宽度对应的信号。 接下来,将脉冲宽度测量电路的输出输入到一个比较器和另一个比较器。 一个比较器鉴别脉冲宽度等于或大于第一设定值的时间,另一个比较器鉴别脉冲宽度等于或小于第二设定值的时间。 比较器连接到上/下计数器的上端和下端。 上/下计数器的输出通过减法器连接到峰化滤波器,根据升/降计数器的计数值控制峰值滤波器的声学特性。
    • 8. 发明申请
    • ACOUSTIC CHARACTERISTIC CONTROL APPARATUS
    • 声学特性控制装置
    • US20110140770A1
    • 2011-06-16
    • US13057896
    • 2009-08-03
    • Yasushi SatoAtsuko Ryu
    • Yasushi SatoAtsuko Ryu
    • H03K5/00
    • G10L25/48G10H1/0008G10H2210/061G10L25/09
    • An acoustic characteristic control apparatus supplies music signal, for example, to input terminal connected to a band-pass filter and a peaking filter. In a zero-cross detection circuit, a pulse signal corresponding to a period while a signal is positive is formed. A pulse-width measuring circuit output a signal corresponding to a pulse width. Next, the output of the pulse-width measuring circuit is inputted to one comparator and another comparator. The one comparator discriminates a time when the pulse width is equal to or larger than a first setting value, and the another comparator discriminates a time when the pulse width is equal to or smaller than a second setting value. The comparator is connected to the up terminal and the down terminal of an up/down counter. The output of the up/down counter is connected to the peaking filter through the subtractor, and acoustic characteristics of the peaking filter is controlled according to the count value of the up/down counter.
    • 声学特性控制装置例如将音乐信号提供给连接到带通滤波器和峰值滤波器的输入端子。 在零交叉检测电路中,形成与信号为正的周期相对应的脉冲信号。 脉冲宽度测量电路输出与脉冲宽度对应的信号。 接下来,将脉冲宽度测量电路的输出输入到一个比较器和另一个比较器。 一个比较器鉴别脉冲宽度等于或大于第一设定值的时间,另一个比较器鉴别脉冲宽度等于或小于第二设定值的时间。 比较器连接到上/下计数器的上端和下端。 上/下计数器的输出通过减法器连接到峰化滤波器,根据升/降计数器的计数值控制峰值滤波器的声学特性。
    • 9. 发明申请
    • ADAPTIVE DIFFERENTIAL PULSE CODE MODULATION ENCODING APPARATUS AND DECODING APPARATUS
    • 自适应差分脉冲编码调制编码装置和解码装置
    • US20110260893A1
    • 2011-10-27
    • US13142010
    • 2009-12-25
    • Yasushi SatoAtsuko Ryu
    • Yasushi SatoAtsuko Ryu
    • H03M7/34
    • G10L19/04H03M7/3046
    • A signal corresponding to a short-period change and a signal corresponding to a long-period change of a sound signal are detected, and optimal quantization is performed based on the combination of the two signals. In an ADPCM encoding apparatus (100), a differential value dn between a 16-bit input signal Xn and a decoded signal Yn-1 of one sample ago is calculated by a subtractor (102). Thereafter, the 16-bit differential value dn is adaptively quantized by an adaptive quantizing section (103), so as to be converted to a (1 to 8)-bit length-variable ADPCM value Dn. Thereafter, the ADPCM value Dn is compression-encoded by a compression-encoding section (108) to generate a signal D′n, and the signal D′n is framed by a framing section (130) and outputted. Further, in an ADPCM decoding apparatus, a framed input signal is subjected to a reverse of the aforesaid process so as to be decoded.
    • 检测对应于短周期变化的信号和对应于声音信号的长周期变化的信号,并且基于两个信号的组合来执行最佳量化。 在ADPCM编码装置(100)中,通过减法器(102)计算16位输入信号Xn与一个采样前的解码信号Yn-1之间的微分值dn。 此后,通过自适应量化部分(103)自适应地量化16位微分值dn,以便将其转换为(1至8)位长度可变ADPCM值Dn。 此后,ADPCM值Dn由压缩编码部(108)进行压缩编码,生成信号D'n,信号D'n由成帧部(130)构成并输出。 此外,在ADPCM解码装置中,对帧输入信号进行与上述处理相反的处理,以进行解码。
    • 10. 发明申请
    • HIGH-FREQUENCY SIGNAL INTERPOLATION APPARATUS AND HIGH-FREQUENCY SIGNAL INTERPOLATION METHOD
    • 高频信号插值装置和高频信号插值方法
    • US20100057230A1
    • 2010-03-04
    • US12448419
    • 2007-12-25
    • Yasushi SatoAtsuko Ryu
    • Yasushi SatoAtsuko Ryu
    • G06F17/00
    • G10L21/038
    • A favorable high-frequency signal is generated and practical high-frequency signal interpolation is implemented through simple processing. A digital audio signal reproduced by an instrument, which also carries out compression, is supplied as an original signal to an input terminal 1, and this original signal is then supplied to a digital sample and hold circuit 3 via a band-pass filter 2. The signal from the digital sample and hold circuit 3 is supplied to a ±1 multiplier 6, which then alternately inverts sign bits. The harmonic components of this signal in which the sign bits are inverted alternately are extracted by a high-pass filter (HPF) 7. Meanwhile, the original signal from the input terminal 1 is supplied to a delay circuit 8 equivalent to the processing time consumed by the aforementioned digital sample and hold circuit 3 and related circuits, forming an adjusted, delayed signal. The signals from the high-pass filter (HPF) 7 and the delay circuit 8 are then added by an adder 10, and the resulting added signal is then output to an output terminal 11.
    • 产生有利的高频信号,通过简单的处理实现了实用的高频信号插值。 由进行压缩的仪器再现的数字音频信号作为原始信号提供给输入端1,然后该原始信号经由带通滤波器2被提供给数字采样和保持电路3。 来自数字采样和保持电路3的信号被提供给±1乘法器6,然后交替地反转符号位。 通过高通滤波器(HPF)7提取符号位交替反转的该信号的谐波分量。同时,来自输入端1的原始信号被提供给等效于所消耗的处理时间的延迟电路8 通过上述数字采样保持电路3和相关电路,形成经调整的延迟信号。 来自高通滤波器(HPF)7和延迟电路8的信号然后由加法器10相加,然后将所得到的相加信号输出到输出端子11。