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1. 发明授权

US09319791B2 Reduced-delay subband signal processing system and method 有权
标题翻译：减速延迟子带信号处理系统及方法
公开(公告)号：US09319791B2
公开(公告)日：2016-04-19
申请号：US13840302
申请日：2013-03-15
申请人： Yair Kerner
发明人： Yair Kerner
IPC分类号： H03G5/00 , H04B3/20 , H04B15/00 , H04R3/04 , G10L21/0232 , H04M9/08 , G10L21/0208
CPC分类号： G10L21/0232 , G10L21/038 , G10L2021/02082 , H03H17/0213 , H04B3/20 , H04M9/082 , H04R3/04 , H04R2430/03
摘要： A method for signal processing, receiving a time domain signal having a sample-rate Fs and generating N time domain signal bands, each having a bandwidth equal to Fs/N. Receiving the N signal bands and transforming a first time domain signal band to a frequency domain at a first resolution and a second time domain signal band to the frequency domain at a second resolution, where the first resolution may be different from the second resolution. Determining one or more first filter coefficients using the frequency domain components from the first signal band and one or more second filter coefficients using the frequency domain components from the second signal band. Transforming the first and second filter coefficients from the frequency domain to a time domain. Applying the first and second time domain filter coefficients to the first and second time domain signals, respectively.
摘要翻译：一种用于信号处理的方法，接收具有采样率Fs并产生N个时域信号频带的时域信号，每个具有等于Fs / N的带宽。接收N个信号频带并以第一分辨率和第二时域信号频带将第一时域信号频带变换到频域，以第二分辨率以第二分辨率转换到频域，其中第一分辨率可以不同于第二分辨率。使用来自第一信号频带的频域分量和使用来自第二信号频带的频域分量的一个或多个第二滤波器系数来确定一个或多个第一滤波器系数。将第一和第二滤波器系数从频域转换到时域。分别将第一和第二时域滤波器系数应用于第一和第二时域信号。

2. 发明授权

US08656415B2 Method and system for removal of clicks and noise in a redirected audio stream 有权
标题翻译：用于消除重定向音频流中的点击和噪音的方法和系统
公开(公告)号：US08656415B2
公开(公告)日：2014-02-18
申请号：US12165590
申请日：2008-06-30
申请人： James W. Wihardja , Xiaoyan Vivian Qian , Jonathan Chien , Yair Kerner
发明人： James W. Wihardja , Xiaoyan Vivian Qian , Jonathan Chien , Yair Kerner
IPC分类号： G06F3/00 , H03G3/00
CPC分类号： G10L21/0208 , G10L19/025 , G10L21/0232
摘要： There is provided a method for use in a computer for redirecting an audio stream from a first audio endpoint to a second audio endpoint in the computer having a controller and a computer operating system for execution by the controller. The method includes directing the audio stream from a client application running on the computer operating system through a first audio resource stack of the computer to the first audio endpoint. The method further includes creating an audio endpoint bridge using the computer operating system, where the audio endpoint bridge provides a path for the audio stream to be directed from the first audio resource stack to pass through a second audio resource stack to the second audio endpoint. The method may also enhance the audio stream to reduce noise and redirect the audio stream to the second audio endpoint using the audio endpoint bridge.
摘要翻译：提供了一种在计算机中用于将音频流从第一音频端点重定向到具有控制器的计算机中的第二音频端点的方法，以及用于由控制器执行的计算机操作系统。该方法包括将来自计算机操作系统上运行的客户机应用的音频流引导到计算机的第一音频资源堆栈到第一音频端点。该方法还包括使用计算机操作系统创建音频端点桥接器，其中音频端点桥接器提供用于使音频流从第一音频资源栈引导以将第二音频资源栈传递到第二音频端点的路径。该方法还可以增强音频流以减少噪声，并使用音频端点桥将音频流重定向到第二音频端点。

3. 发明授权

US08243952B2 Microphone array calibration method and apparatus 有权
标题翻译：麦克风阵列校准方法和装置
公开(公告)号：US08243952B2
公开(公告)日：2012-08-14
申请号：US12341777
申请日：2008-12-22
申请人： Trausti Thormundsson , Harry K. Lau , Yair Kerner
发明人： Trausti Thormundsson , Harry K. Lau , Yair Kerner
IPC分类号： H04R3/00
CPC分类号： H04R3/005
摘要： An apparatus for providing real-time calibration for two or more microphones. A calibrator for receiving a left microphone signal and a right microphone signal and generating phase difference data. A phase and amplitude correction system for receiving one of the left microphone signal or the right microphone signal the phase difference data and generating calibration data for a beamformer. The beamformer receiving the calibration data, the left microphone signal and the right microphone signal and generating a monaural beamformed signal.
摘要翻译：一种用于为两个或更多麦克风提供实时校准的装置。用于接收左麦克风信号和右麦克风信号并产生相位差数据的校准器。用于接收左麦克风信号或右麦克风信号之一的相位和幅度校正系统向相位差数据发出信号并产生波束形成器的校准数据。波束形成器接收校准数据，左麦克风信号和右麦克风信号，并产生单声道波束形成信号。

4. 发明授权

US06731726B1 Communication on hold 有权
标题翻译：沟通暂停
公开(公告)号：US06731726B1
公开(公告)日：2004-05-04
申请号：US09592707
申请日：2000-06-13
申请人： Yair Kerner , Iddit Shalem , Sverrir Olafsson
发明人： Yair Kerner , Iddit Shalem , Sverrir Olafsson
IPC分类号： H04M1100
CPC分类号： H04L5/1438 , H04L29/06 , H04L69/24 , H04M11/06
摘要： A communication session is established between an APCM modem and a DPCM modem. The central office interrupts the communication session by transmitting a call-waiting alert signal to the APCM modem. In response to the alert signal, the APCM modem transmits a caller ID data request to the central office. The APCM modem further transmits an on hold notification to the DPCM modem to prevent the DPCM modem from terminating the communication session as a result of the interruption by the alert signal and/or receipt of the caller ID data by the APCM modem. The DPCM modem is placed on hold while the APCM modem answers the call waiting. Subsequently, the APCM modem initiates a quick connect procedure to re-establish the communication session with the DPCM modem.
摘要翻译：在APCM调制解调器和DPCM调制解调器之间建立通信会话。中央局通过向APCM调制解调器发送呼叫等待警报信号来中断通信会话。响应于警报信号，APCM调制解调器向中心局发送呼叫者ID数据请求。 APCM调制解调器进一步向DPCM调制解调器发送保留通知，以防止DPCM调制解调器由于警报信号的中断和/或APCM调制解调器接收到呼叫者ID数据而终止通信会话。 APCM调制解调器应答呼叫等待时，DPCM调制解调器处于保持状态。随后，APCM调制解调器启动快速连接过程以重新建立与DPCM调制解调器的通信会话。

5. 发明申请

US20120308040A1 MICROPHONE ARRAY CALIBRATION METHOD AND APPARATUS 审中-公开
标题翻译：麦克风阵列校准方法和装置
公开(公告)号：US20120308040A1
公开(公告)日：2012-12-06
申请号：US13585836
申请日：2012-08-15
申请人： Trausti Thormundsson , Harry K. Lau , Yair Kerner
发明人： Trausti Thormundsson , Harry K. Lau , Yair Kerner
IPC分类号： H04R3/00
CPC分类号： H04R3/005
摘要： An apparatus for providing real-time calibration for two or more microphones. A calibrator for receiving a left microphone signal and a right microphone signal and generating phase difference data. A phase and amplitude correction system for receiving one of the left microphone signal or the right microphone signal the phase difference data and generating calibration data for a beamformer. The beamformer receiving the calibration data, the left microphone signal and the right microphone signal and generating a monaural beamformed signal.
摘要翻译：一种用于为两个或更多麦克风提供实时校准的装置。用于接收左麦克风信号和右麦克风信号并产生相位差数据的校准器。用于接收左麦克风信号或右麦克风中的一个的相位和幅度校正系统向相位差数据发送信号并产生波束形成器的校准数据。波束形成器接收校准数据，左麦克风信号和右麦克风信号，并产生单声道波束形成信号。

6. 发明申请

US20110200195A1 SYSTEMS AND METHODS FOR SPEAKER BAR SOUND ENHANCEMENT 有权
标题翻译：扬声器声音增强的系统和方法
公开(公告)号：US20110200195A1
公开(公告)日：2011-08-18
申请号：US13092006
申请日：2011-04-21
申请人： Harry K. Lau , Yair Kerner , Jonathan Chien , James W. Wihardja
发明人： Harry K. Lau , Yair Kerner , Jonathan Chien , James W. Wihardja
IPC分类号： H04R5/00
CPC分类号： H04S7/30 , H04S3/002 , H04S2400/01
摘要： A sound reproduction apparatus comprising a loudspeaker bar having a plurality of loudspeakers. A loudspeaker bar controller coupled to the loudspeaker bar for processing audio data for the plurality of loudspeakers, the loudspeaker bar controller comprising a spatial enhancement/virtualization system for receiving a surround channel of audio data and processing the surround channel of audio data with a spatial generation/virtualization filter, wherein a left stereo channel of audio data and a right stereo channel of audio data are not processed with the head related transfer function filter. Bass is enhanced for small speakers which are not able to produce bass frequencies.
摘要翻译：一种声音再现装置，包括具有多个扬声器的扬声器条。扬声器控制器，耦合到扬声器条，用于处理多个扬声器的音频数据，扬声器条控制器包括空间增强/虚拟化系统，用于接收音频数据的环绕声道并用空间生成处理音频数据的环绕声道 /虚拟化过滤器，其中音频数据的左立体声通道和音频数据的右立体声通道不被头部相关传递函数滤波器处理。对于不能产生低音频率的小型扬声器，低音增强。

7. 发明申请

US20090086987A1 Method and System for Removal of Clicks and Noise in a Redirected Audio Stream 有权
标题翻译：重定向音频流中点击和噪声的方法和系统
公开(公告)号：US20090086987A1
公开(公告)日：2009-04-02
申请号：US12165590
申请日：2008-06-30
申请人： James W. Wihardja , Xiaoyan Vivian Qian , Jonathan Chien , Yair Kerner
发明人： James W. Wihardja , Xiaoyan Vivian Qian , Jonathan Chien , Yair Kerner
IPC分类号： G10K11/16 , H04B3/00
CPC分类号： G10L21/0208 , G10L19/025 , G10L21/0232
摘要： There is provided a method of redirecting an audio stream from a first audio endpoint to a second audio endpoint in a computer operating system. The method includes directing the audio stream from a client application through a first audio resource stack to the first audio endpoint; creating an audio endpoint bridge to provide a path for the audio stream from the first audio resource stack through a second audio resource stack connected to the second audio endpoint; and redirecting the audio stream to the second audio endpoint using the audio endpoint bridge. The audio endpoint bridge can be created by forming a bridging application so as to activate the second audio stack. The bridging application can be hooked into a Windows audio engine in the second audio resource stack. The bridge can be used to intercept an audio stream and remove noise from it. Additionally, a specific type of noise with sporadic intermittent spikes has been observed in this audio framework with certain Bluetooth headsets. A system and method is described for removing this specific type of noise.
摘要翻译：提供了一种将音频流从第一音频端点重定向到计算机操作系统中的第二音频端点的方法。该方法包括将音频流从客户应用通过第一音频资源堆栈引导到第一音频端点; 创建音频端点桥接器，以提供来自第一音频资源堆栈的音频流通过连接到第二音频端点的第二音频资源栈的路径; 并使用音频端点桥将音频流重定向到第二音频端点。音频端点桥可以通过形成桥接应用来激活第二音频栈来创建。桥接应用程序可以连接到第二个音频资源堆栈中的Windows音频引擎。该桥可用于拦截音频流并从中消除噪音。此外，在具有某些蓝牙耳机的音频框架中已经观察到具有零星间歇尖峰的特定类型的噪声。描述了用于去除这种特定类型的噪声的系统和方法。

8. 发明申请

US20130266158A1 CLASS-D AMPLIFIER WITH PULSE DENSITY MODULATION OUTPUT FEEDBACK FOR HIGHER PERFORMANCE ACOUSTIC ECHO CANCELLER 有权
标题翻译：具有脉冲密度调制输出反馈的高分辨率放大器用于更高性能的声学ECHO CANCELLER
公开(公告)号：US20130266158A1
公开(公告)日：2013-10-10
申请号：US13860477
申请日：2013-04-10
申请人： Mouna Elkhatib , Trausti Thormundsson , Jonathan Chien , Yair Kerner
发明人： Mouna Elkhatib , Trausti Thormundsson , Jonathan Chien , Yair Kerner
IPC分类号： H04R3/12
CPC分类号： H04R3/12 , H03F3/2175 , H04R3/02 , H04R2499/15
摘要： A system for processing audio data comprising an amplifier configured to receive an audio signal and to perform nonlinear processing on the audio signal. An encoder coupled to the amplifier, the encoder configured to receive the nonlinearly processed audio signal and to encode the nonlinearly processed audio signal into a data transmission format. A transmitter configured to receive and transmit the encoded nonlinearly processed audio signal. A receiver configured to receive the transmitted encoded nonlinearly processed audio signal and to decode the encoded nonlinearly processed audio signal. A digital voice processor configured to receive the nonlinearly processed audio signal and to use the nonlinearly processed audio signal for echo estimation and to subsequently subtract the estimated echo signal from a microphone signal.
摘要翻译：一种用于处理音频数据的系统，包括被配置为接收音频信号并对音频信号执行非线性处理的放大器。耦合到放大器的编码器，编码器被配置为接收非线性处理的音频信号并且将非线性处理的音频信号编码为数据传输格式。发射器，被配置为接收和发送经编码的非线性处理的音频信号。接收器，被配置为接收所发送的经编码的非线性处理的音频信号并解码编码的非线性处理的音频信号。一种数字语音处理器，被配置为接收非线性处理的音频信号，并使用非线性处理的音频信号进行回波估计，并随后从麦克风信号中减去估计的回波信号。

9. 发明申请

US20120016501A1 WAVEFORM SHAPING SYSTEM TO PREVENT ELECTRICAL AND MECHANICAL SATURATION IN LOUD SPEAKERS 有权
标题翻译：波形成形系统，以防止扬声器中的电气和机械饱和度
公开(公告)号：US20120016501A1
公开(公告)日：2012-01-19
申请号：US13183288
申请日：2011-07-14
申请人： Trausti Thormundsson , Govind Kannan , Shlomi I. Regev , James W. Wihardja , Yair Kerner , Harry K. Lau , Ragnar H. Jonsson
发明人： Trausti Thormundsson , Govind Kannan , Shlomi I. Regev , James W. Wihardja , Yair Kerner , Harry K. Lau , Ragnar H. Jonsson
IPC分类号： G06F17/00
CPC分类号： H03G7/007
摘要： Peak reduction and power limitations are used to prevent distortion and protect components. In a cellular telephone, peak reduction can be based on battery power level to prevent electrical distortion from saturation. In addition peak reduction can be used to prevent mechanical distortion such as rub and buzz. Dynamic range compression can be used for peak reduction. In another application dynamic range compression can be used to control the power output to protect a speaker from damage. One example of a dynamic range compressor/peak limiter comprises a look-ahead buffer and an analysis engine. For example, the look-ahead buffer holds a window of samples of a signal. The analysis engine selects a gain envelope function on the basis of the samples, for example, by selecting the Pth sample in the buffer whenever that sample exceeds a given threshold.
摘要翻译：峰值降低和功率限制用于防止失真并保护组件。在蜂窝电话中，峰值降低可以基于电池功率电平，以防止电气失真饱和。此外，还可以使用峰值降低来防止机械变形，例如摩擦和嗡嗡声。动态范围压缩可用于降低峰值。在另一个应用中，可以使用动态范围压缩来控制功率输出，以保护扬声器免受损坏。动态范围压缩器/峰值限制器的一个示例包括预先提前缓冲器和分析引擎。例如，先行缓冲器保存信号样本的窗口。分析引擎基于样本选择增益包络函数，例如，每当该样本超过给定阈值时，通过选择缓冲器中的第P个样本。

10. 发明申请

US20110170683A1 Systems and Methods for Echo Cancellation and Echo Suppression 失效
标题翻译：回波消除和回波抑制的系统和方法
公开(公告)号：US20110170683A1
公开(公告)日：2011-07-14
申请号：US12684829
申请日：2010-01-08
申请人： Youhong Lu , Trausti Thormundsson , Yair Kerner , Ragnar H. Jonsson
发明人： Youhong Lu , Trausti Thormundsson , Yair Kerner , Ragnar H. Jonsson
IPC分类号： H04M9/08
CPC分类号： H04M3/002 , H04M9/082
摘要： Traditionally, echo cancellation has employed linear adaptive filters to cancel echoes in a two way communication system. The rate of adaptation is often dynamic and varies over time. Disclosed are novel rates of adaptation that perform well in the presence of background noise, during double talk and with echo path changes. Additionally, the echo or residual echo can further be suppressed with non-linear processing performed using joint frequency-time domain processing.
摘要翻译：传统上，回波消除采用线性自适应滤波器来消除双向通信系统中的回波。适应率通常是动态的，随时间而变化。公开了在双重通话期间和具有回波路径改变的情况下，在背景噪声的存在下表现良好的适应性适应性。此外，通过使用联合频域处理执行的非线性处理，可进一步抑制回波或残留回波。

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