公开(公告)号：US08380497B2

公开(公告)日：2013-02-19

申请号：US12579322

申请日：2009-10-14

申请人： Asif I. Mohammad ,  Dinesh Ramakrishnan

发明人： Asif I. Mohammad ,  Dinesh Ramakrishnan

IPC分类号： G10L21/02

CPC分类号： G10L25/78 ,  G10L21/0216 ,  G10L25/48

摘要： A system and method are disclosed for noise level/spectrum estimation and speech activity detection. Some embodiments include a probabilistic model to estimate noise level and subsequently detect the presence of speech. These embodiments outperform standard voice activity detectors (VADs), producing improved detection in a variety of noisy environments.

摘要翻译： 公开了用于噪声水平/频谱估计和语音活动检测的系统和方法。 一些实施例包括估计噪声水平并随后检测语音存在的概率模型。 这些实施例优于标准语音活动检测器（VAD），在各种噪声环境中产生改进的检测。

公开(公告)号：US08811601B2

公开(公告)日：2014-08-19

申请号：US13079548

申请日：2011-04-04

申请人： Asif I. Mohammad ,  Kwokleung Chan ,  Ian Ernan Liu ,  Erik Visser ,  Dinesh Ramakrishnan

发明人： Asif I. Mohammad ,  Kwokleung Chan ,  Ian Ernan Liu ,  Erik Visser ,  Dinesh Ramakrishnan

IPC分类号： H04M9/08 ,  H04B3/23

CPC分类号： H04B3/234 ,  G10L21/0208 ,  G10L2021/02082 ,  G10L2021/02166 ,  H04M9/082

摘要： A method for echo cancellation and noise suppression is disclosed. Linear echo cancellation (LEC) is performed for a primary microphone channel on an entire frequency band or in a range of frequencies where echo is audible. LEC is performed on one or more secondary microphone channels only on a lower frequency range over which spatial processing is effective. The microphone channels are spatially processed over the lower frequency range after LEC. Non-linear noise suppression post-processing is performed on the entire frequency band. Echo post-processing is performed on the entire frequency band.

摘要翻译： 公开了一种用于回波消除和噪声抑制的方法。 线性回波消除（LEC）是在整个频带上的主要麦克风通道或在可听见回声的频率范围内执行的。 只有在空间处理有效的较低频率范围上的一个或多个辅助麦克风通道上执行LEC。 麦克风通道在LEC之后的较低频率范围内进行空间处理。 在整个频带上执行非线性噪声抑制后处理。 在整个频带上执行回波后处理。

公开(公告)号：US20120250882A1

公开(公告)日：2012-10-04

申请号：US13079548

申请日：2011-04-04

申请人： Asif I. Mohammad ,  Kwokleung Chan ,  Ian Eman Liu ,  Erik Visser ,  Dinesh Ramakrishnan

发明人： Asif I. Mohammad ,  Kwokleung Chan ,  Ian Eman Liu ,  Erik Visser ,  Dinesh Ramakrishnan

IPC分类号： H04B15/00

CPC分类号： H04B3/234 ,  G10L21/0208 ,  G10L2021/02082 ,  G10L2021/02166 ,  H04M9/082

摘要： A method for echo cancellation and noise suppression is disclosed. Linear echo cancellation (LEC) is performed for a primary microphone channel on an entire frequency band or in a range of frequencies where echo is audible. LEC is performed on one or more secondary microphone channels only on a lower frequency range over which spatial processing is effective. The microphone channels are spatially processed over the lower frequency range after LEC. Non-linear noise suppression post-processing is performed on the entire frequency band. Echo post-processing is performed on the entire frequency band.

摘要翻译： 公开了一种用于回波消除和噪声抑制的方法。 线性回波消除（LEC）是在整个频带上的主要麦克风通道或在可听见回声的频率范围内执行的。 只有在空间处理有效的较低频率范围上的一个或多个辅助麦克风通道上执行LEC。 麦克风通道在LEC之后的较低频率范围内进行空间处理。 在整个频带上执行非线性噪声抑制后处理。 在整个频带上执行回波后处理。

公开(公告)号：US20100094625A1

公开(公告)日：2010-04-15

申请号：US12579322

申请日：2009-10-14

申请人： Asif I. Mohammad ,  Dinesh Ramakrishnan

发明人： Asif I. Mohammad ,  Dinesh Ramakrishnan

IPC分类号： G10L15/20

CPC分类号： G10L25/78 ,  G10L21/0216 ,  G10L25/48

摘要： A system and method are disclosed for noise level/spectrum estimation and speech activity detection. Some embodiments include a probabilistic model to estimate noise level and subsequently detect the presence of speech. These embodiments outperform standard voice activity detectors (VADs), producing improved detection in a variety of noisy environments.

摘要翻译： 公开了用于噪声水平/频谱估计和语音活动检测的系统和方法。 一些实施例包括估计噪声水平并随后检测语音存在的概率模型。 这些实施例优于标准语音活动检测器（VAD），在各种噪声环境中产生改进的检测。

公开(公告)号：US08411880B2

公开(公告)日：2013-04-02

申请号：US12022052

申请日：2008-01-29

申请人： Song Wang ,  Dinesh Ramakrishnan ,  Eddie L. T. Choy

发明人： Song Wang ,  Dinesh Ramakrishnan ,  Eddie L. T. Choy

IPC分类号： H03F99/00

CPC分类号： H04R3/005 ,  H03G3/32 ,  H04M1/0214 ,  H04M1/035 ,  H04M1/6008 ,  H04R29/006 ,  H04R2410/05 ,  H04R2499/11

摘要： Sound signal reception is improved by utilizing a plurality of microphones to capture sound signals which are then weighed to dynamically adjust signal quality. A first sound signal and a second sound signal are obtained from first and second microphones, respectively, where the first and second sound signals originate from one or more sound sources. A first signal characteristic (e.g., signal power, signal signal-to-noise ratio, etc.) is obtained for the first sound signal and a second signal characteristic is obtained for the second sound signal. The first and second sound signals are weighed or scaled based on their respective first and second signal characteristics. The weighed first and second sound signals are then combined to obtain an output sound signal.

摘要翻译： 通过利用多个麦克风来捕获声音信号来改善声音信号接收，然后称重声音信号以动态地调整信号质量。 从第一和第二麦克风分别获得第一声音信号和第二声音信号，其中第一和第二声音信号源自一个或多个声源。 获得第一声音信号的第一信号特性（例如，信号功率，信号信噪比等），并且获得第二声音信号的第二信号特性。 第一和第二声音信号基于它们各自的第一和第二信号特性被称重或缩放。 然后将称重的第一和第二声音信号组合以获得输出声音信号。

公开(公告)号：US20090190774A1

公开(公告)日：2009-07-30

申请号：US12022037

申请日：2008-01-29

申请人： Song Wang ,  Dinesh Ramakrishnan ,  Samir Gupta ,  Eddie L.T. Choy

发明人： Song Wang ,  Dinesh Ramakrishnan ,  Samir Gupta ,  Eddie L.T. Choy

IPC分类号： H04R3/00

CPC分类号： G10L21/028 ,  G10L2021/02166 ,  H04R3/005 ,  H04R25/40

摘要： An enhanced blind source separation technique is provided to improve separation of highly correlated signal mixtures. A beamforming algorithm is used to precondition correlated first and second input signals in order to avoid indeterminacy problems typically associated with blind source separation. The beamforming algorithm may apply spatial filters to the first signal and second signal in order to amplify signals from a first direction while attenuating signals from other directions. Such directionality may serve to amplify a desired speech signal in the first signal and attenuate the desired speech signal from the second signal. Blind source separation is then performed on the beamformer output signals to separate the desired speech signal and the ambient noise and reconstruct an estimate of the desired speech signal. To enhance the operation of the beamformer and/or blind source separation, calibration may be performed at one or more stages.

摘要翻译： 提供增强的盲源分离技术来改善高度相关的信号混合物的分离。 波束形成算法用于预处理相关的第一和第二输入信号，以避免通常与盲源分离相关联的不确定性问题。 波束成形算法可以对第一信号和第二信号应用空间滤波器，以便在衰减来自其它方向的信号的同时放大来自第一方向的信号。 这种方向性可以用于在第一信号中放大期望的语音信号，并从第二信号中衰减所需的语音信号。 然后对波束形成器输出信号执行盲源分离，以分离所需的语音信号和环境噪声，并重构所需语音信号的估计。 为了增强波束形成器和/或盲源分离的操作，可以在一个或多个阶段执行校准。

公开(公告)号：US09552840B2

公开(公告)日：2017-01-24

申请号：US13280303

申请日：2011-10-24

申请人： Lae-Hoon Kim ,  Erik Visser ,  Pei Xiang ,  Ian Ernan Liu ,  Dinesh Ramakrishnan

发明人： Lae-Hoon Kim ,  Erik Visser ,  Pei Xiang ,  Ian Ernan Liu ,  Dinesh Ramakrishnan

IPC分类号： H04R3/00 ,  H04R1/02 ,  H04R9/06 ,  G11B20/00

CPC分类号： G11B20/00 ,  H04R3/005 ,  H04R2201/401 ,  H04S2400/15

摘要： A method for audio signal processing is described. The method includes decomposing a recorded auditory scene into a first category of localizable sources and a second category of ambient sound. The method also includes recording an indication of the directions of each of the localizable sources. The method may be performed with a device having a microphone array.

摘要翻译： 描述用于音频信号处理的方法。 该方法包括将记录的听觉场景分解为第一类可本地化的源和第二类环境声音。 该方法还包括记录每个可本地化的源的方向的指示。 该方法可以用具有麦克风阵列的设备来执行。

公开(公告)号：US08571231B2

公开(公告)日：2013-10-29

申请号：US12782147

申请日：2010-05-18

申请人： Dinesh Ramakrishnan ,  Homayoun Shahri ,  Song Wang

发明人： Dinesh Ramakrishnan ,  Homayoun Shahri ,  Song Wang

IPC分类号： H04B15/00

CPC分类号： G10L21/0208 ,  G10L21/0232

摘要： An electronic device for suppressing noise in an audio signal is described. The electronic device includes a processor and instructions stored in memory. The electronic device receives an input audio signal and computes an overall noise estimate based on a stationary noise estimate, a non-stationary noise estimate and an excess noise estimate. The electronic device also computes an adaptive factor based on an input Signal-to-Noise Ratio (SNR) and one or more SNR limits. A set of gains is also computed using a spectral expansion gain function. The spectral expansion gain function is based on the overall noise estimate and the adaptive factor. The electronic device also applies the set of gains to the input audio signal to produce a noise-suppressed audio signal and provides the noise-suppressed audio signal.

摘要翻译： 描述了用于抑制音频信号中的噪声的电子设备。 电子设备包括处理器和存储在存储器中的指令。 电子设备接收输入音频信号，并且基于静态噪声估计，非平稳噪声估计和过量噪声估计来计算总体噪声估计。 电子设备还基于输入信噪比（SNR）和一个或多个SNR限制来计算自适应因子。 还使用频谱扩展增益函数来计算一组增益。 频谱扩展增益函数基于总噪声估计和自适应因子。 电子设备还将该组增益应用于输入音频信号以产生噪声抑制的音频信号并提供噪声抑制的音频信号。

公开(公告)号：US09113240B2

公开(公告)日：2015-08-18

申请号：US12405057

申请日：2009-03-16

申请人： Dinesh Ramakrishnan ,  Song Wang

发明人： Dinesh Ramakrishnan ,  Song Wang

IPC分类号： H04B15/00 ,  H04R3/00 ,  G10L21/028 ,  G10L21/0216 ,  G10L21/0308 ,  H04R29/00

CPC分类号： H04R3/005 ,  G10L21/028 ,  G10L21/0308 ,  G10L2021/02165 ,  G10L2021/02166 ,  H04R29/006 ,  H04R2420/07 ,  H04R2430/03 ,  H04R2430/20 ,  H04R2499/11

摘要： Signal processing solutions take advantage of microphones located on different devices and improve the quality of transmitted voice signals in a communication system. With usage of various devices such as Bluetooth headsets, wired headsets and the like in conjunction with mobile handsets, multiple microphones located on different devices are exploited for improving performance and/or voice quality in a communication system. Audio signals are recorded by microphones on different devices and processed to produce various benefits, such as improved voice quality, background noise reduction, voice activity detection and the like.

摘要翻译： 信号处理解决方案利用位于不同设备上的麦克风，并提高通信系统中传输的语音信号的质量。 随着诸如蓝牙耳机，有线耳机等各种设备的使用，与移动手机结合使用，位于不同设备上的多个麦克风被用来改善通信系统中的性能和/或语音质量。 音频信号由不同设备上的麦克风记录，并被处理以产生各种益处，例如改进的语音质量，背景噪声降低，语音活动检测等。

公开(公告)号：US08650238B2

公开(公告)日：2014-02-11

申请号：US11946253

申请日：2007-11-28

申请人： Dinesh Ramakrishnan ,  Song Wang ,  Eddie L. T. Choy ,  Samir Kumar Gupta

发明人： Dinesh Ramakrishnan ,  Song Wang ,  Eddie L. T. Choy ,  Samir Kumar Gupta

IPC分类号： G06F7/38

CPC分类号： H04J3/0632 ,  G10L19/005

摘要： In a digital system with more than one clock source, lack of synchronization between the clock sources may cause overflow or underflow in sample buffers, also called sample slipping. Sample slipping may lead to undesirable artifacts in the processed signal due to discontinuities introduced by the addition or removal of extra samples. To smooth out discontinuities caused by sample slipping, samples are filtered to when a buffer overflow condition occurs, and the samples are interpolated to produce additional samples when a buffer underflow condition occurs. The interpolated samples may also be filtered. The filtering and interpolation operations can be readily implemented without adding significant burden to the computational complexity of a real-time digital system.

摘要翻译： 在具有多个时钟源的数字系统中，时钟源之间的同步缺乏可能导致采样缓冲器中的溢出或下溢，也称为样品打滑。 由于添加或除去额外的样品引起的不连续性，样品打滑可能导致处理过的信号中的不期望的伪影。 为了平滑由样品滑动引起的不连续性，将样品过滤到发生缓冲液溢出状态时，当发生缓冲液下溢条件时，样品被内插以产生附加样品。 内插样本也可以被过滤。 可以容易地实现滤波和插值操作，而不会对实时数字系统的计算复杂度造成重大负担。