公开(公告)号：US08831932B2

公开(公告)日：2014-09-09

申请号：US13294471

申请日：2011-11-11

Applicant: Jinwei Feng ,  Peter L. Chu ,  Stephen Botzko

Inventor： Jinwei Feng ,  Peter L. Chu ,  Stephen Botzko

IPC: G10L11/04

CPC classification number: G10L19/002 ,  G10L19/0212 ,  G10L19/24 ,  G10L25/18 ,  H04M3/568 ,  H04M7/0072

Abstract: Use of a scalable audio codec to implement distributed mixing and/or sender bit rate regulation in a multipoint conference is disclosed. The scalable audio codec allows the audio signal from each endpoint to be split into one or more frequency bands and for the transform coefficients within such bands to be prioritized such that usable audio may be decoded from a subset of the entire signal. The subset may be created by omitting certain frequency bands and/or by omitting certain coefficients within the frequency bands. By providing various rules for each endpoint in a conference, the endpoint can determine the importance of its signal to the conference and can select an appropriate bit rate, thereby conserving bandwidth and/or processing power throughout the conference.

Abstract translation: 公开了使用可扩展音频编解码器来实现多点会议中的分布式混合和/或发送器比特率调节。 可扩展音频编解码器允许来自每个端点的音频信号被分成一个或多个频带，并且为了使这些频带内的变换系数被优先化，使得可以从整个信号的子集中解码可用的音频。 可以通过省略某些频带和/或通过省略频带内的某些系数来创建该子集。 通过为会议中的每个端点提供各种规则，端点可以确定其信号到会议的重要性，并且可以选择适当的比特率，从而节省整个会议的带宽和/或处理能力。

公开(公告)号：US20110069830A1

公开(公告)日：2011-03-24

申请号：US12565374

申请日：2009-09-23

Applicant: Peter L. Chu ,  Wayne Dunlap

Inventor： Peter L. Chu ,  Wayne Dunlap

IPC: H04M9/08

CPC classification number: H04M3/002 ,  H04M1/6033 ,  H04M3/2236 ,  H04M3/568 ,  H04M9/085

Abstract: Audio from a near-end that has been acoustically coupled at the far-end and returned to the near-end unit is detected and suppressed at the near-end of a conference. First and second energy outputs for separate bands are determined for the near-end audio being sent from the near-end unit and for the far-end audio being received at the near-end unit. The near-end unit compares the first and second energy outputs to one another for each of the bands over a time delay range and detects the return of the sent near-end audio in the received far-end audio based on the comparison. The comparison can use a cross-correlation to find an estimated time delay used for further analysis of the near and far-end energies. The near-end unit suppresses any detected return by muting or reducing what far-end audio is output at its loudspeaker.

Abstract translation: 来自远端的声耦合并返回到近端单元的近端的音频在会议的近端被检测和抑制。 确定用于单独频带的第一和第二能量输出用于从近端单元发送的近端音频以及在近端单元处接收的远端音频。 近端单元在时间延迟范围内针对每个频带将第一和第二能量输出彼此进行比较，并且基于该比较来检测所接收的远端音频中所发送的近端音频的返回。 比较可以使用互相关来找到用于进一步分析近端和远端能量的估计时间延迟。 近端单元通过静音抑制任何检测到的返回，或者减少在其扬声器处输出远端音频。

公开(公告)号：US07787328B2

公开(公告)日：2010-08-31

申请号：US11015373

申请日：2004-12-17

Applicant: Peter L. Chu ,  Michael Kenoyer ,  Richard Washington

Inventor： Peter L. Chu ,  Michael Kenoyer ,  Richard Washington

IPC: G01S3/802

CPC classification number: H04R3/005 ,  G01S5/18 ,  H04N7/142 ,  H04R2201/401 ,  H04R2430/23

Abstract: In accordance with the present invention, a system and method for computing a location of an acoustic source is disclosed. The method includes steps of processing a plurality of microphone signals in frequency space to search a plurality of candidate acoustic source locations for a maximum normalized signal energy. The method uses phase-delay look-up tables to efficiently determine phase delays for a given frequency bin number k based upon a candidate source location and a microphone location, thereby reducing system memory requirements. Furthermore, the method compares a maximum signal energy for each frequency bin number k with a threshold energy Et(k) to improve accuracy in locating the acoustic source.

Abstract translation: 根据本发明，公开了一种用于计算声源的位置的系统和方法。 该方法包括在频率空间中处理多个麦克风信号以搜索多个候选声源位置以获得最大归一化信号能量的步骤。 该方法使用相位延迟查找表来有效地确定基于候选源位置和麦克风位置的给定频率仓数k的相位延迟，从而减少系统存储器要求。 此外，该方法将每个频率仓数k的最大信号能量与阈值能量Et（k）进行比较，以提高定位声源的精度。

公开(公告)号：US06593956B1

公开(公告)日：2003-07-15

申请号：US09079840

申请日：1998-05-15

Applicant: Steven L. Potts ,  Hong Wang ,  Wendi Beth Rabiner ,  Peter L. Chu

Inventor： Steven L. Potts ,  Hong Wang ,  Wendi Beth Rabiner ,  Peter L. Chu

IPC: H04N714

CPC classification number: G01S3/7865 ,  G01S3/8083 ,  H04N7/15

Abstract: A system, such as a video conferencing system, is provided which includes an image pickup device, an audio pickup device, and an audio source locator. The image pickup device generates image signals representative of an image, while the audio pickup device generates audio signals representative of sound from an audio source, such as speaking person. The audio source locator processes the image signals and audio signals to determine a direction of the audio source relative to a reference point. The system can further determine a location of the audio source relative to the reference point. The reference point can be a camera. The system can use the direction or location information to frame a proper camera shot which would include the audio source.

Abstract translation: 提供了诸如视频会议系统的系统，其包括图像拾取装置，音频拾取装置和音频源定位器。 图像拾取装置产生表示图像的图像信号，而音频拾取装置生成表示来自诸如说话人之类的音频源的声音的音频信号。 音频源定位器处理图像信号和音频信号以确定音频源相对于参考点的方向。 系统可以进一步确定音频源相对于参考点的位置。 参考点可以是相机。 该系统可以使用方向或位置信息来构成包括音频源的正确的相机拍摄。

公开(公告)号：US5715319A

公开(公告)日：1998-02-03

申请号：US657636

申请日：1996-05-30

Applicant: Peter L. Chu

Inventor： Peter L. Chu

IPC: H04R1/40 ,  H04R3/00 ,  H04R5/027

CPC classification number: H04R3/005 ,  H04R2201/401 ,  H04R2201/403 ,  H04R2201/405 ,  H04R25/407

Abstract: An end fire microphone array having reduced analog-to-digital converter requirements is disclosed. Analog filters are used to band-limit at least two secondary microphone elements which are spaced from a primary microphone element a distance respective of their band limited outputs. The band-limited secondary microphone outputs are combined by an analog summer and the primary microphone and combined secondary microphone signals are digitized by an analog-to-digital converter. A signal processor performs a super-directive analysis of the primary microphone signal and the combined secondary microphone signals. A steerable superdirective microphone array is disclosed. A plurality of microphones are arranged in a ring. The microphone outputs are digitized, split into frequency bands, and weighted sums are formed for each of a plurality of directions. A steering control circuit evaluates the relative energy of each directional signal in each band and selects a microphone direction for further processing and output.

Abstract translation: 公开了一种具有降低的模数转换器要求的终端式麦克风阵列。 模拟滤波器用于限制至少两个辅助麦克风元件，这些次要麦克风元件与主麦克风元件间隔开它们的频带限制输出的距离。 频带限制的二次麦克风输出通过模拟加法器组合，主麦克风和组合次级麦克风信号由模数转换器数字化。 信号处理器执行主麦克风信号和组合次级麦克风信号的超指令分析。 公开了一种可操纵的超导麦克风阵列。 多个麦克风布置在环中。 麦克风输出被数字化，分成频带，并且为多个方向中的每一个形成加权和。 转向控制电路评估每个频带中每个定向信号的相对能量，并选择麦克风方向进行进一步处理和输出。

公开(公告)号：US5263019A

公开(公告)日：1993-11-16

申请号：US837729

申请日：1992-02-19

Applicant: Peter L. Chu

Inventor： Peter L. Chu

IPC: H04J1/00 ,  H04B3/23 ,  H04M9/08 ,  H04R3/02

CPC classification number: H04R3/02 ,  H04M9/082 ,  H04R2430/03

Abstract: An improved echo cancelling device for reducing the effects of acoustic feedback between a loudspeaker and microphone in a communication system. The device includes an adjustable filter for receiving a loudspeaker signal and generating in response thereto an echo estimation signal. The device subtracts the echo estimation signal from the microphone signal to produce an echo corrected microphone signal. During periods of time when the microphone signal is substantially derived from acoustic feedback between the loudspeaker and the microphone, the device adjusts transfer characteristics of the filter to reduce the echo corrected microphone signal. The improvement includes estimating from the adjusted transfer characteristics an energy transfer ratio representative of the ratio of the energy of the microphone signal to the energy of the loudspeaker signal. The device compares the microphone signal to the energy transfer ratio multiplied by the loudspeaker signal to identify periods of time when the microphone signal is substantially derived from acoustic feedback between the loudspeaker and the microphone.

Abstract translation: 一种用于减少通信系统中扬声器和麦克风之间的声反馈的影响的改进的回波消除装置。 该装置包括可调滤波器，用于接收扬声器信号并响应于此产生回波估计信号。 该装置从麦克风信号中减去回波估计信号，以产生回波校正麦克风信号。 在麦克风信号基本上来源于扬声器和麦克风之间的声反馈的时间段期间，设备调节滤波器的传送特性以减少回波校正的麦克风信号。 该改进包括从调整的传送特性估计表示麦克风信号的能量与扬声器信号的能量的比率的能量传递比。 该装置将麦克风信号与乘以扬声器信号的能量传递比进行比较，以识别麦克风信号基本上从扬声器和麦克风之间的声反馈导出的时间段。

公开(公告)号：US08625776B2

公开(公告)日：2014-01-07

申请号：US12565374

申请日：2009-09-23

Applicant: Peter L. Chu ,  Wayne Dunlap

Inventor： Peter L. Chu ,  Wayne Dunlap

IPC: H04M9/08

CPC classification number: H04M3/002 ,  H04M1/6033 ,  H04M3/2236 ,  H04M3/568 ,  H04M9/085

Abstract: Audio from a near-end that has been acoustically coupled at the far-end and returned to the near-end unit is detected and suppressed at the near-end of a conference. First and second energy outputs for separate bands are determined for the near-end audio being sent from the near-end unit and for the far-end audio being received at the near-end unit. The near-end unit compares the first and second energy outputs to one another for each of the bands over a time delay range and detects the return of the sent near-end audio in the received far-end audio based on the comparison. The comparison can use a cross-correlation to find an estimated time delay used for further analysis of the near and far-end energies. The near-end unit suppresses any detected return by muting or reducing what far-end audio is output at its loudspeaker.

Abstract translation: 来自远端的声耦合并返回到近端单元的近端的音频在会议的近端被检测和抑制。 确定用于单独频带的第一和第二能量输出用于从近端单元发送的近端音频以及在近端单元处接收的远端音频。 近端单元在时间延迟范围内针对每个频带将第一和第二能量输出彼此进行比较，并且基于该比较来检测所接收的远端音频中所发送的近端音频的返回。 比较可以使用互相关来找到用于进一步分析近端和远端能量的估计时间延迟。 近端单元通过静音抑制任何检测到的返回，或者减少在其扬声器处输出远端音频。

公开(公告)号：US20130106977A1

公开(公告)日：2013-05-02

申请号：US13282609

申请日：2011-10-27

Applicant: Peter L. Chu ,  Yibo Liu

Inventor： Peter L. Chu ,  Yibo Liu

IPC: H04N7/15 ,  H04M3/42

CPC classification number: H04N7/15 ,  G06F3/165 ,  H04M3/568 ,  H04N7/142

Abstract: A videoconferencing system has a videoconferencing unit that use portable devices as peripherals for the system. The portable devices obtain near-end audio and send the audio to the videoconferencing unit via a wireless connection. In turn, the videoconferencing unit sends the near-end audio from the loudest portable device along with near-end video to the far-end. The portable devices can control the videoconferencing unit and can initially establish the videoconference by connecting with the far-end and then transferring operations to the videoconferencing unit. To deal with acoustic coupling between the unit's loudspeaker and the portable device's microphone, the unit uses an echo canceller that is compensated for differences in the clocks used in the A/D and D/A converters of the loudspeaker and microphone.

Abstract translation: 视频会议系统具有视频会议单元，其使用便携式设备作为系统的外围设备。 便携式设备获得近端音频，并通过无线连接将音频发送到视频会议单元。 反过来，视频会议单元将最接近便携式设备的近端音频和近端视频一起发送到远端。 便携式设备可以控制视频会议单元，并且可以通过与远端连接然后将操作传送到视频会议单元来最初建立视频会议。 为了处理单元的扬声器和便携式设备的麦克风之间的声耦合，该单元使用回波消除器，其补偿扬声器和麦克风的A / D和D / A转换器中使用的时钟的差异。

公开(公告)号：US08219400B2

公开(公告)日：2012-07-10

申请号：US12275393

申请日：2008-11-21

Applicant: Peter L. Chu

Inventor： Peter L. Chu

IPC: G10L11/00

CPC classification number: H04R27/00 ,  H04M3/56

Abstract: Stereo to mono voice conferencing conversion is performed during a voice conference. Conferencing equipment receives audio for right and left channels and filters each of the channels into a plurality of bands. For each band of each channel, the equipment determines an energy level and compares each energy level for each band of the right channel to each energy level for each corresponding band of the left channel. Based on the comparison, the equipment determines which channel has more audio resulting from speech. Based on the determination, the equipment adjusts delivery of the audio from the right and left channels to a mono channel for transmission to endpoints only capable of mono audio in the voice conference.

Abstract translation: 在语音会议期间执行立体声到单声道语音会议转换。 会议设备接收用于右声道和左声道的音频，并且将每个声道滤波成多个频带。 对于每个通道的每个频带，设备确定能量水平，并将每个频带的每个频带的每个能级与每个相应的左声道频带的能级进行比较。 基于比较，设备确定哪个频道具有更多的来自语音的音频。 基于确定，设备将音频从右声道和左声道的传送调整到单声道，以传输到只能在语音会议中单声道音频的端点。

公开(公告)号：US08130977B2

公开(公告)日：2012-03-06

申请号：US11320323

申请日：2005-12-27

Applicant: Peter L. Chu

Inventor： Peter L. Chu

IPC: H04R3/00 ,  H04R1/02 ,  H04R29/00 ,  H04R5/00

CPC classification number: H04R3/005 ,  H04R1/406

Abstract: An arbitrarily positioned cluster of three microphones can be used for stereo input of a videoconferencing system. To produce stereo input, right and left weightings for signal inputs from each of the microphones are determined. The right and left weightings correspond to preferred directive patterns for stereo input of the system. The determined right weightings are applied to the signal inputs from each of the microphones, and the weighted inputs are summed to product the right input. The same is done for the left input using the determined left weightings. The three microphones are preferably first-order, cardioid microphone capsules spaced close together in an audio unit, where each faces radially outward at 120-degrees. The orientation of the arbitrarily positioned cluster relative to the system can be determined by directly detecting the orientation or by using stored arrangements.

Abstract translation: 三个麦克风的任意定位的群集可以用于视频会议系统的立体声输入。 为了产生立体声输入，确定来自每个麦克风的信号输入的右和右加权。 右和右加权对应于系统的立体声输入的优选指令模式。 确定的权重被应用于来自每个麦克风的信号输入，并且将加权输入相加以产生正确的输入。 对于左输入，使用确定的左权重也是一样。 这三个麦克风最好是在音频单元中彼此靠近的一级的心形麦克风胶囊，其中每个都以120度径向向外。 可以通过直接检测取向或使用存储的布置来确定任意定位的簇相对于系统的取向。