公开(公告)号：US20060245601A1

公开(公告)日：2006-11-02

申请号：US11116117

申请日：2005-04-27

Applicant: Francois Michaud ,  Jean-Marc Valin ,  Jean Rouat

Inventor： Francois Michaud ,  Jean-Marc Valin ,  Jean Rouat

IPC: H04R3/00

CPC classification number: H04R3/005 ,  G01S5/22 ,  H04R2201/403

Abstract: The present invention relates to a system for localizing at least one sound source, comprising a set of spatially spaced apart sound sensors to detect sound from the at least one sound source and produce corresponding sound signals, and a frequency-domain beamformer responsive to the sound signals from the sound sensors and steered in a range of directions to localize, in a single step, the at least one sound source. The present invention is also concerned with a system for tracking a plurality of sound sources, comprising a set of spatially spaced apart sound sensors to detect sound from the sound sources and produce corresponding sound signals, and a sound source particle filtering tracker responsive to the sound signals from the sound sensors for simultaneously tracking the plurality of sound sources. The invention still further relates to a system for localizing and tracking a plurality of sound sources, comprising a set of spatially spaced apart sound sensors to detect sound from the sound sources and produce corresponding sound signals; a sound source detector responsive to the sound signals from the sound sensors and steered in a range of directions to localize the sound sources, and a particle filtering tracker connected to the sound source detector for simultaneously tracking the plurality of sound sources.

Abstract translation: 本发明涉及一种用于定位至少一个声源的系统，包括一组空间间隔开的声音传感器，以检测来自至少一个声源的声音并产生相应的声音信号，以及响应于声音的频域波束形成器 来自声音传感器的信号并且在一个方向范围内被引导以在一个步骤中定位至少一个声源。 本发明还涉及用于跟踪多个声源的系统，包括一组空间间隔开的声音传感器以检测来自声源的声音并产生相应的声音信号，以及响应于声音的声源粒子滤波跟踪器 来自声音传感器的信号，用于同时跟踪多个声源。 本发明还涉及一种用于定位和跟踪多个声源的系统，包括一组空间间隔开的声音传感器以检测来自声源的声音并产生对应的声音信号; 声源检测器，其响应于来自声音传感器的声音信号并在一定方向上被引导以定位声源;以及粒子滤波跟踪器，连接到声源检测器，用于同时跟踪多个声源。

公开(公告)号：US20050281415A1

公开(公告)日：2005-12-22

申请号：US11197817

申请日：2005-08-05

Applicant: Russell Lambert ,  Shi-Ping Hsu ,  Karina Edmonds

Inventor： Russell Lambert ,  Shi-Ping Hsu ,  Karina Edmonds

IPC: B60R16/02 ,  B60R11/02 ,  H04B3/20 ,  H04R1/40 ,  H04R3/00 ,  H04R1/02

CPC classification number: H04R3/005 ,  H04R2201/403

Abstract: Apparatus and a corresponding method for processing speech signals in a noisy reverberant environment, such as an automobile. An array of microphones (10) receives speech signals from a relatively fixed source (12) and noise signals from multiple sources (32) reverberated over multiple paths. One of the microphones is designated a reference microphone and the processing system includes adaptive frequency impulse response (FIR) filters (24) enabled by speech detection circuitry (21) and coupled to the other microphones to align their output signals with the reference microphone output signal. The filtered signals are then combined in a summation circuit (18). Signal components derived from the speech signal combine coherently in the summation circuit, while noise signal components combine incoherently, resulting in composite output signal with an improved signal-to-noise ratio. The composite output signal is further processed in a speech conditioning circuit (20) to reduce the effects of reverberation.

Abstract translation: 用于在噪声混响环境（例如汽车）中处理语音信号的装置和相应方法。 麦克风阵列（10）从相对固定的源（12）接收语音信号，并且从多个源（32）的多个路径上混响的噪声信号。 麦克风中的一个被指定为参考麦克风，并且处理系统包括由语音检测电路（21）启用的自适应频率脉冲响应（FIR）滤波器（24），并且耦合到另一麦克风以将其输出信号与参考麦克风输出信号 。 滤波的信号然后在求和电路（18）中组合。 来自语音信号的信号分量在求和电路中相干地组合，而噪声信号分量非均匀地组合，导致具有改善的信噪比的复合输出信号。 复合输出信号在语音调理电路（20）中被进一步处理，以减少混响的影响。

公开(公告)号：US20050276423A1

公开(公告)日：2005-12-15

申请号：US09937022

申请日：2001-09-19

Applicant: Roland Aubauer ,  Ralf Kern ,  Stefano Klinke ,  Dieter Leckschat

Inventor： Roland Aubauer ,  Ralf Kern ,  Stefano Klinke ,  Dieter Leckschat

IPC: H04R3/00 ,  H04R29/00 ,  H04R1/02 ,  H03G3/20

CPC classification number: H04R29/006 ,  H04R3/005 ,  H04R2201/403

Abstract: In order to record and to process audio signals with a good useful-signal to noise-signal ratio in acoustic noise conditions and with a good ratio between the direct sound and the reflected sound in an environment which in particular has no reverberation electrical signals produced by conversion of audio signals recorded by a predetermined microphone arrangement are processed in such a manner that, if the sound pressure levels at the microphones in the microphone arrangement are the same, electrical signals which are produced by these microphones but are of different intensity—different sensitivities of the microphones—are automatically matched, without any manual matching procedures needed to be carried out individually and separately. A microphone arrangement is based on combining the characteristics of an array of microphones with those of a method for matching the sensitivity of microphones.

Abstract translation: 为了在声学噪声条件下记录和处理具有良好的有用信号与噪声信号比的音频信号，并且在环境中具有良好的直接声音和反射声音之间的比例，该环境中特别没有混响电信号 由预定麦克风装置记录的音频信号的转换以如下方式进行处理，即如果麦克风装置中麦克风的声压级相同，则由这些麦克风产生但是具有不同强度不同灵敏度的电信号 的麦克风自动匹配，无需单独和单独执行任何手动匹配程序。 麦克风布置基于将麦克风阵列的特征与用于匹配麦克风的灵敏度的方法的特征组合。

公开(公告)号：US06792118B2

公开(公告)日：2004-09-14

申请号：US10004141

申请日：2001-11-14

Applicant: Lloyd Watts

Inventor： Lloyd Watts

IPC: H04R300

CPC classification number: H04R3/005 ,  H04R2201/403

Abstract: Determining a time delay between a first signal received at a first sensor and a second signal received at a second sensor is described. The first signal is analyzed to derive a plurality of first signal channels at different frequencies and the second signal is analyzed to derive a plurality of second signal channels at different frequencies. A first feature detected that occurs at a first time in one of the first signal channels. A second feature is detected that occurs at a second time in one of the second signal channels. The first feature is matched with the second feature and the first time is compared to the second time to determine the time delay.

Abstract translation: 确定在第一传感器处接收的第一信号与在第二传感器处接收的第二信号之间的时间延迟。 分析第一信号以导出不同频率的多个第一信号信道，并且分析第二信号以导出不同频率的多个第二信号信道。 检测到的第一特征是在第一信号信道之一中第一次发生。 检测在第二信号信道之一中第二次发生的第二特征。 第一个功能与第二个功能相匹配，第一次与第二次比较确定时间延迟。

公开(公告)号：US20030157965A1

公开(公告)日：2003-08-21

申请号：US10258670

申请日：2003-03-20

Inventor： Claude Marro ,  Francois Hamon

IPC: H04B001/38

CPC classification number: H04R3/005 ,  H01Q3/2605 ,  H04B7/086 ,  H04R2201/403

Abstract: The present invention relates to a receiving system for a multi-sensor antenna and comprising: at least one set of channel filters (310i, 710k; inull1 . . . N) filtering the signals Xi(t, f) received by the different antenna sensors, where these signals may include a desired signal (Si(t, f)); a summer (350, 720) summing the filtered signals nullVi(t, f)null, where inull1 . . . N, by means of the channel filters and emitting an antenna output signal Y(t,f), at least one calculating module (320, 720) receiving either the sensor signals and aligning them in phase or the signals filtered by the channel filters, said module estimating the transfer function V(t, f), W k(t, f)) of an optimum filter in a manner to minimize the square difference between the antenna output signals filtered by said filter and the desired signal characterized in that it furthermore comprises at least one statistical analysis module (330, 730) operating on the frequency values of the transfer function.

Abstract translation: 多传感器天线的接收系统技术领域本发明涉及一种用于多传感器天线的接收系统，包括：至少一组信道滤波器（310i，710k; i = 1，...，N），对由不同的传感器接收的信号Xi（t，f）进行滤波 天线传感器，其中这些信号可以包括期望的信号（Si（t，f））; 一个夏天（350,720）对滤波后的信号[Vi（t，f）]求和，其中i = 1。 。 。 N，通过信道滤波器并发射天线输出信号Y（t，f），至少一个计算模块（320,720）接收传感器信号并将它们相位对准或由信道滤波器滤波的信号， 所述模块以最小化由所述滤波器滤波的天线输出信号与期望信号之间的平方差的方式估计最佳滤波器的传递函数V（t，f），W k（t，f）），其特征在于， 还包括对传递函数的频率值进行操作的至少一个统计分析模块（330,730）。

公开(公告)号：US06603858B1

公开(公告)日：2003-08-05

申请号：US09088694

申请日：1998-06-01

Applicant: George Raicevich ,  Harvey Dillon

Inventor： George Raicevich ,  Harvey Dillon

IPC: H03G320

CPC classification number: H04R25/407 ,  H04R3/005 ,  H04R2201/403 ,  H04R2225/43

Abstract: An apparatus and method for processing sound, suitable for use in association with a hearing aid, cochlear implant prosthesis or the like. Coupled to an array of microphones (1) are a pair of fixed array processors (2,4) each having different characteristic signal-to-noise performances and internal noise parameters in different levels of ambient noise. Based upon an ambient noise estimate derived from noise floor detector (8) a control circuit (5) controls the gain of a pair of VCA's (7,9) coupled to the fixed array processors (2,4) in order to produce an output signal from summer (16) which maximises the signal-to-noise ratio of a signal emanating from a source in an on-beam direction relative to the microphone array (10).

Abstract translation: 一种用于处理声音的装置和方法，适用于与助听器，人工耳蜗假体等相关联。 耦合到麦克风阵列（1）的是一对固定阵列处理器（2,4），每个固定阵列处理器具有不同的特征信噪比性能和不同环境噪声水平的内部噪声参数。 基于从噪声本底检测器（8）导出的环境噪声估计，控制电路（5）控制耦合到固定阵列处理器（2,4）的一对VCA（7,9）的增益，以产生输出 来自夏天（16）的信号，其使得相对于麦克风阵列（10）在光束方向上从源发出的信号的信噪比最大化。

公开(公告)号：US20030142836A1

公开(公告)日：2003-07-31

申请号：US09966873

申请日：2001-09-28

Inventor： Daniel Max Warren ,  Stephen C. Thompson

IPC: H04R025/00 ,  H04R003/00

CPC classification number: H04R25/407 ,  H04R1/406 ,  H04R3/005 ,  H04R25/405 ,  H04R29/006 ,  H04R2201/403

Abstract: A directional microphone system is disclosed, which comprises circuitry for low pass filtering a first order signal, and circuitry for high pass filtering a second order signal. The system further comprises circuitry for summing the low pass filtered first order signal and the high pass filtered second order signal. A method of determining whether a plurality of microphones have sufficiently matched frequency response characteristics to be used in a multi-order directional microphone array is also disclosed. For a microphone array having at least three microphones, wherein one of the microphones is disposed between the other of the microphones, a method of determining the arrangement of the microphones in the array is also disclosed.

Abstract translation: 公开了一种定向麦克风系统，其包括用于对第一阶信号进行低通滤波的电路，以及用于对二阶信号进行高通滤波的电路。 该系统还包括用于对低通滤波的一阶信号和高通滤波的二阶信号求和的电路。 还公开了一种确定多个麦克风是否具有足够匹配的频率响应特性以用于多阶定向麦克风阵列的方法。 对于具有至少三个麦克风的麦克风阵列，其中麦克风之一设置在麦克风的另一个之间，还公开了一种确定阵列中的麦克风的布置的方法。

公开(公告)号：US20030095667A1

公开(公告)日：2003-05-22

申请号：US10004141

申请日：2001-11-14

Applicant: Applied Neurosystems Corporation

Inventor： Lloyd Watts

IPC: H04R029/00

CPC classification number: H04R3/005 ,  H04R2201/403

Abstract: Determining a time delay between a first signal received at a first sensor and a second signal received at a second sensor is described. The first signal is analyzed to derive a plurality of first signal channels at different frequencies and the second signal is analyzed to derive a plurality of second signal channels at different frequencies. A first feature detected that occurs at a first time in one of the first signal channels. A second feature is detected that occurs at a second time in one of the second signal channels. The first feature is matched with the second feature and the first time is compared to the second time to determine the time delay.

Abstract translation: 确定在第一传感器处接收的第一信号与在第二传感器处接收的第二信号之间的时间延迟。 分析第一信号以导出不同频率的多个第一信号信道，并且分析第二信号以导出不同频率的多个第二信号信道。 检测到的第一特征是在第一信号信道之一中第一次发生。 检测在第二信号信道之一中第二次发生的第二特征。 第一个功能与第二个功能相匹配，第一次与第二次比较确定时间延迟。

公开(公告)号：US20030026437A1

公开(公告)日：2003-02-06

申请号：US10196318

申请日：2002-07-16

Inventor： Cornelis Pieter Janse ,  Harm Jan Willem Belt

IPC: A61F011/06 ,  G10K011/16 ,  H03B029/00

CPC classification number: H04R3/02 ,  H04R3/005 ,  H04R3/12 ,  H04R27/00 ,  H04R2201/403 ,  H04R2430/20

Abstract: A sound reinforcement system (1) comprises at least one microphone (2), adaptive echo compensation (EC) means (4) coupled to said microphone (2) for generating a microphone signal, and one or more loudspeakers (3) coupled to the EC means (4). In addition it comprises a dynamic echo suppressor (DES 7) coupled between the adaptive EC means (4) and said at least one loudspeaker (3) for suppressing remaining echoes by using a time delay between the amplitudes of a microphones signal frequency component and the same remaining echo frequency component. Echo emanating from a room wherein the listener resides is effectively removed, and even a fine tuned model can effectively be made in cases wherein the speaker(s) move. The sound reinforcement system (1), which may be a hands-free system is embodied as a public address system, a congress system, a conferencing system, or a communication system such as a passenger communication system for a vehicle such as a car, aeroplane or the like.

Abstract translation: 扩音系统（1）包括至少一个麦克风（2），耦合到所述麦克风（2）的自适应回波补偿（EC）装置（4），用于产生麦克风信号，以及一个或多个耦合到所述麦克风的扬声器 EC意味着（4）。 此外，它包括耦合在自适应EC装置（4）和所述至少一个扬声器（3）之间的动态回声抑制器（DES 7），用于通过使用麦克风信号频率分量的振幅和 相同的剩余回波频率分量。 从听众居住的房间发出的回声被有效地移除，并且甚至在扬声器移动的情况下也可以有效地进行微调模型。 可以是免提系统的扩声系统（1）被体现为公共广播系统，会议系统，会议系统或诸如汽车等车辆的乘客通信系统的通信系统， 飞机等。

公开(公告)号：US20020064287A1

公开(公告)日：2002-05-30

申请号：US09983284

申请日：2001-10-23

Inventor： Takashi Kawamura ,  Takeo Kanamori ,  Toshihide Ishimoto

IPC: H04R003/00 ,  H04B015/00

CPC classification number: H04R3/005 ,  H04R1/406 ,  H04R2201/401 ,  H04R2201/403

Abstract: A pickup section 11 transduces sounds to audio signals. A zoom control section 12 outputs a zoom position signal corresponding to a zoom position. A directivity control section 13 alters the directivity characteristics under telescopic operation so as to mainly pickup sounds coming from a frontal direction with an enhancement which is in accordance with the zoom position signal, thereby outputting an R channel audio signal and an L channel audio signal. In accordance with the zoom position signal, a noise suppression section 14 applies a greater degree of suppression to the background noise contained in the respective channel audio signals under telescopic operation than under wide-angle operation. As a result, a target sound from a remote location can be picked up with a sufficient enhancement in accordance with the zoom position under telescopic operation.

Abstract translation: 拾音部分11将声音转换成音频信号。 变焦控制部12输出与变焦位置对应的变焦位置信号。 方向性控制部分13改变伸缩操作下的方向性特征，主要是从正面方向进行拾音，具有与变焦位置信号相应的增强，从而输出R声道音频信号和L声道音频信号。 根据变焦位置信号，噪声抑制部分14对广播角度操作下的各个声道音频信号中包含的背景噪声进行更大的抑制。 结果，可以根据伸缩操作下的变焦位置以足够的增强来拾取来自远程位置的目标声音。