会员体验
专利管家(专利管理)
工作空间(专利管理)
风险监控(情报监控)
数据分析(专利分析)
侵权分析(诉讼无效)
联系我们
交流群
官方交流:
QQ群: 891211   
微信请扫码    >>>
现在联系顾问~
热词
    • 5. 发明授权
    • Audio beamforming
    • 音频波束成形
    • US09084037B2
    • 2015-07-14
    • US13384720
    • 2010-07-22
    • Rene Martinus Maria Derkx
    • Rene Martinus Maria Derkx
    • H04R29/00H04R3/00
    • H04R3/005H04R2201/401H04R2430/25
    • An audio beamforming apparatus includes a receiving circuit (103) which receives signals from an at least two-dimensional microphone array (101). A reference circuit (105) generates reference beams and a combining circuit (107) generates an output signal corresponding to a desired beam pattern by combining the reference beams. An estimation circuit (109) generates a direction estimate by determining angles corresponding to local minima for a power measure of the output signal in at least a first and respectively second angle interval. The direction estimate is generated by selecting one of the angles. The combining circuit (107) determines combination parameters to provide a notch in an angle corresponding to the direction estimate and a maximization of a directivity cost measure where the directivity cost measure is indicative of a ratio between a gain in the first direction and an energy averaged gain.
    • 音频波束成形装置包括从至少二维麦克风阵列(101)接收信号的接收电路(103)。 参考电路(105)产生参考光束,并且组合电路(107)通过组合参考光束来产生对应于期望光束图案的输出信号。 估计电路(109)通过在至少第一和第二角度间隔中确定与输出信号的功率测量值相对应的与局部最小值相对应的角度来产生方向估计。 通过选择一个角度来生成方向估计。 组合电路(107)确定组合参数以提供与方向估计相对应的角度的陷波和方向性成本测量的最大化,其中方向性成本度量指示第一方向上的增益与平均的能量之间的比率 获得。
    • 6. 发明授权
    • Noise reduction of breathing signals
    • 呼吸信号降噪
    • US08834386B2
    • 2014-09-16
    • US13382304
    • 2010-07-02
    • Rene Martinus Maria Derkx
    • Rene Martinus Maria Derkx
    • A61B5/08A61B7/00
    • A61B7/003A61B5/0803
    • The invention relates to a system for and a method of processing breathing signals. A noise reduction operation is performed on a spectral breathing signal (18) to compute an output spectral signal (38), said noise reduction operation using spectral subtraction; and a two-dimensional frequency and time filtering (32) of a gain function (30) used in the spectral subtraction of the noise reduction operation performing step is performed, for example, a two-dimensional frequency and time median filtering of the gain function. For example, said spectral breathing signal is computed based on a breathing signal.
    • 本发明涉及一种处理呼吸信号的系统和方法。 对频谱呼吸信号(18)进行噪声降低操作以计算输出频谱信号(38),使用频谱减法的所述降噪操作; 并且执行在降噪操作执行步骤的谱减法中使用的增益函数(30)的二维频率和时间滤波(32),例如增益函数的二维频率和时间中值滤波 。 例如,基于呼吸信号计算所述频谱呼吸信号。
    • 7. 发明申请
    • SIGNAL PROCESSING SYSTEM AND METHOD
    • 信号处理系统和方法
    • US20100020984A1
    • 2010-01-28
    • US12513529
    • 2007-11-08
    • Cornelis Pieter JanseRene Martinus Maria DerkxMarie-Bernadette Gennotte
    • Cornelis Pieter JanseRene Martinus Maria DerkxMarie-Bernadette Gennotte
    • H04B15/00
    • H04R3/02H04R2499/13
    • A signal processing system comprises a microphone (20), a subtractor (22) arranged to receive an output of the microphone (20), an amplifier G arranged to receive an output of the subtractor (22), a rear loudspeaker (24) arranged to receive an output of the amplifier G, a front loudspeaker (26) arranged to receive an output of the amplifier G, and one or more summers (28) interposed between the amplifier G and a loudspeaker (24, 26), the or each summer (28) arranged to add an audio signal m[k] to the signal s[k] received from the amplifier G. The system also includes a mixing matrix D arranged to receive the respective inputs R, F of the rear and front loudspeakers (24, 26) and arranged to output a summation signal R+F and a difference signal R−F, and an adaptive filter SAF; MCAF arranged to receive the outputs R+F, R−F of the mixing matrix D, the subtractor (22) arranged to receive an output of the adaptive filter SAF; MCAF and an output of the subtractor (22) arranged to control the adaptive filter SAF; MCAF.
    • 信号处理系统包括麦克风(20),布置成接收麦克风(20)的输出的减法器(22),布置成接收减法器(22)的输出的放大器G,布置成 以接收放大器G的输出,布置成接收放大器G的输出的前置扬声器(26)和插在放大器G和扬声器(24,26)之间的一个或多个加法器(28),所述或每个 夏季(28)被布置为向从放大器G接收的信号s [k]添加音频信号m [k]。该系统还包括混合矩阵D,其被布置成接收后置和前置扬声器的相应输入R,F (24,26),并且被配置为输出求和信号R + F和差分信号RF,以及自适应滤波器SAF; MCAF被布置为接收混合矩阵D的输出R + F,R-F,减法器(22)被布置为接收自适应滤波器SAF的输出; MCAF和用于控制自适应滤波器SAF的减法器(22)的输出; MCAF。
    • 9. 发明申请
    • SPEECH DETECTOR
    • 语音探测器
    • US20110288864A1
    • 2011-11-24
    • US12950711
    • 2010-11-19
    • Patrick KechichianCornelis Pieter JanseRene Martinus Maria DerkxWouter Joos Tirry
    • Patrick KechichianCornelis Pieter JanseRene Martinus Maria DerkxWouter Joos Tirry
    • G10L15/00
    • H04R3/005G10L25/78G10L2021/02166
    • A method for detecting speech using a first microphone adapted to produce a first signal (x), and a second microphone adapted to produce a second signal (x2), the method comprising the steps of: (i) applying gain to the second signal to produce a normalised second signal, which signal is normalised relative to the first signal; (ii) constructing one or more signal components from the first signal and the normalised second signal; (iii) constructing an adaptive differential microphone (ADM) having a constructed microphone response constructed from the one or more signal components which response has at least one directional null; (iv) producing one or more ADM outputs (yf, yb) from the constructed microphone response in response to detected sound; (v) computing a ratio of a parameter of either a first signal component or a constructed microphone response to a parameter of an output of the ADM; (vi) comparing the ratio to an adaptive threshold value; (vii) detecting speech if the ratio is greater than or equal to the adaptive threshold value.
    • 一种用于使用适于产生第一信号(x)的第一麦克风来检测语音的方法,以及适于产生第二信号(x2)的第二麦克风,所述方法包括以下步骤:(i)将增益应用于第二信号 产生归一化的第二信号,该信号相对于第一信号被归一化; (ii)从第一信号和归一化的第二信号构造一个或多个信号分量; (iii)构造具有由所述一个或多个信号分量构成的构造的麦克风响应的自适应差分麦克风(ADM),所述响应具有至少一个方向空值; (iv)响应于检测到的声音从构成的麦克风响应产生一个或多个ADM输出(yf,yb); (v)计算第一信号分量或构造的麦克风响应的参数与ADM的输出的参数的比率; (vi)将该比率与自适应阈值进行比较; (vii)如果比率大于或等于自适应阈值,则检测语音。
    • 10. 发明申请
    • METHOD OF, AND APPARATUS FOR, PLANAR AUDIO TRACKING
    • 平面音频跟踪的方法和装置
    • US20110264249A1
    • 2011-10-27
    • US13141856
    • 2009-12-21
    • Rene Martinus Maria DerkxCornelis Pieter Janse
    • Rene Martinus Maria DerkxCornelis Pieter Janse
    • G06F17/00
    • H04R1/406G01S3/8083G10L21/0208G10L2021/02166H04R3/005H04R2201/401
    • A planar audio tracking system comprises a square array of four microphones (M1, M2, M3, M4) arranged as first and second cross-dipole microphones and a virtually constructed monopole microphone. The signals from these microphones undergo directional pre-processing and the results are applied to a filtered sum beamformer (FSB) (32). The FSB identifies functions (hd (0), hd (π/2), and hm) of the FSB which are representative of impulse responses from desired audio source(s) to the first and second cross-dipole and the monopole microphone, respectively. The functions of the first cross-dipole and the monopole microphones and the functions of the second cross-dipole and the monopole microphones are cross correlated to produce respective estimates (ψc(l) and ψs(l)) representative of the lag of the most dominant audio source. An angle-estimate ({circumflex over (φ)}) of the most dominant source is determined using the estimates of lag. Other embodiments of the tracking system may comprise 3 microphones arranged in a circular array and forming first and second cross-dipoles and a virtual monopole.
    • 平面音频跟踪系统包括布置为第一和第二交叉偶极麦克风的四个麦克风(M1,M2,M3,M4)的正方形阵列和虚拟构造的单极麦克风。 来自这些麦克风的信号经过定向预处理,并将结果应用于滤波的和波束形成器(FSB)(32)。 FSB标识表示从所需音频源到第一和第二交叉偶极子和单极麦克风的脉冲响应的FSB的功能(hd(0),hd(&pgr; / 2)和hm) 分别。 第一交叉偶极子和单极麦克风的功能和第二交叉偶极子和单极麦克风的功能是交叉相关的,以产生代表最大滞后的相应估计(ψc(1)和ψs(l)) 优势音源 使用滞后的估计来确定最主要的来源的角度估计({circumflex over(&phgr;)})。 跟踪系统的其他实施例可以包括以圆形阵列排列并形成第一和第二交叉偶极子和虚拟单极子的3个麦克风。