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。

公开(公告)号：US20090135976A1

公开(公告)日：2009-05-28

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： H04L7/027

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。

公开(公告)号：US08175871B2

公开(公告)日：2012-05-08

申请号：US11864906

申请日：2007-09-28

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

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

IPC分类号： G10L21/02 ,  G10L19/00 ,  G10L17/00 ,  H04B15/00 ,  H04M9/08

CPC分类号： H04M9/082 ,  G10L2021/02082 ,  G10L2021/02166

摘要： Multiple microphone noise suppression apparatus and methods are described herein. The apparatus and methods implement a variety of noise suppression techniques and apparatus that can be selectively applied to signals received using multiple microphones. The microphone signals received at each of the multiple microphones can be independently processed to cancel echo signal components that can be generated from a local audio source. The echo cancelled signals may be processed by some or all modules within a signal separator that operates to separate or otherwise isolate a speech signal from noise signals. The signal separator can include a pre-processing de-correlator followed by a blind source separator. The output of the blind source separator can be post filtered to provide post separation de-correlation. The separated speech and noise signals can be non-linearly processed for further noise reduction, and additional post processing can be implemented following the non-linear processing.

摘要翻译： 本文描述了多麦克风噪声抑制装置和方法。 该装置和方法实现了可以选择性地应用于使用多个麦克风接收的信号的各种噪声抑制技术和装置。 可以独立地处理在多个麦克风中的每一个处接收的麦克风信号以消除可从本地音频源产生的回波信号分量。 回波消除信号可以由信号分离器内的一些或所有模块进行处理，信号分离器用于分离或以其他方式将语音信号与噪声信号隔离开来。 信号分离器可以包括预处理去相关器，其后是盲源分离器。 盲源分离器的输出可以后过滤以提供后分离去相关。 分离的语音和噪声信号可以被非线性处理以进一步降低噪声，并且可以在非线性处理之后实现附加的后处理。

公开(公告)号：US20090089054A1

公开(公告)日：2009-04-02

申请号：US11864906

申请日：2007-09-28

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

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

IPC分类号： G10L15/20

CPC分类号： H04M9/082 ,  G10L2021/02082 ,  G10L2021/02166

摘要： Multiple microphone noise suppression apparatus and methods are described herein. The apparatus and methods implement a variety of noise suppression techniques and apparatus that can be selectively applied to signals received using multiple microphones. The microphone signals received at each of the multiple microphones can be independently processed to cancel echo signal components that can be generated from a local audio source. The echo cancelled signals may be processed by some or all modules within a signal separator that operates to separate or otherwise isolate a speech signal from noise signals. The signal separator can include a pre-processing de-correlator followed by a blind source separator. The output of the blind source separator can be post filtered to provide post separation de-correlation. The separated speech and noise signals can be non-linearly processed for further noise reduction, and additional post processing can be implemented following the non-linear processing.

摘要翻译： 本文描述了多麦克风噪声抑制装置和方法。 该装置和方法实现了可以选择性地应用于使用多个麦克风接收的信号的各种噪声抑制技术和装置。 可以独立地处理在多个麦克风中的每一个处接收的麦克风信号以消除可从本地音频源产生的回波信号分量。 回波消除信号可以由信号分离器内的一些或所有模块进行处理，信号分离器用于分离或以其他方式将语音信号与噪声信号隔离开来。 信号分离器可以包括预处理去相关器，其后是盲源分离器。 盲源分离器的输出可以后过滤以提供后分离去相关。 分离的语音和噪声信号可以被非线性处理以进一步降低噪声，并且可以在非线性处理之后实现附加的后处理。

公开(公告)号：US08954324B2

公开(公告)日：2015-02-10

申请号：US11864897

申请日：2007-09-28

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

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

IPC分类号： G10L15/20 ,  G10L11/06 ,  G10L19/00 ,  G10L25/78 ,  G10L21/0216

CPC分类号： G10L25/78 ,  G10L2021/02165

摘要： Voice activity detection using multiple microphones can be based on a relationship between an energy at each of a speech reference microphone and a noise reference microphone. The energy output from each of the speech reference microphone and the noise reference microphone can be determined. A speech to noise energy ratio can be determined and compared to a predetermined voice activity threshold. In another embodiment, the absolute value of the autocorrelation of the speech and noise reference signals are determined and a ratio based on autocorrelation values is determined. Ratios that exceed the predetermined threshold can indicate the presence of a voice signal. The speech and noise energies or autocorrelations can be determined using a weighted average or over a discrete frame size.

摘要翻译： 使用多个麦克风的语音活动检测可以基于语音基准麦克风和噪声参考麦克风各自的能量之间的关系。 可以确定来自每个语音参考麦克风和噪声参考麦克风的能量输出。 可以确定语音能量比，并将其与预定的语音活动阈值进行比较。 在另一个实施例中，确定语音和噪声参考信号的自相关的绝对值，并且确定基于自相关值的比率。 超过预定阈值的比率可以指示语音信号的存在。 可以使用加权平均值或离散的帧大小来确定语音和噪声能量或自相关性。

公开(公告)号：US07970564B2

公开(公告)日：2011-06-28

申请号：US11551509

申请日：2006-10-20

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

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

IPC分类号： G01R13/00

CPC分类号： G06K9/6243 ,  G10L21/0272

摘要： This disclosure describes signal processing techniques that can improve the performance of blind source separation (BSS) techniques. In particular, the described techniques propose pre-processing steps that can help to de-correlate the different signals from one another prior to execution of the BSS techniques. In addition, the described techniques also propose optional post-processing steps that can further de-correlate the different signals following execution of the BSS techniques. The techniques may be particularly useful for improving BSS performance with highly correlated audio signals, e.g., from two microphones that are in close spatial proximity to one another.

摘要翻译： 本公开描述了可以提高盲源分离（BSS）技术的性能的信号处理技术。 特别地，所描述的技术提出了预处理步骤，其可以有助于在执行BSS技术之前将不同信号彼此相关联。 此外，所描述的技术还提出了可选的后处理步骤，其可以在执行BSS技术之后进一步使不同信号去相关。 这些技术对于通过高度相关的音频信号（例如来自彼此紧密地空间接近的两个麦克风）来改善BSS性能可能特别有用。

公开(公告)号：US07528745B2

公开(公告)日：2009-05-05

申请号：US11452836

申请日：2006-06-13

申请人： Song Wang ,  Eddie L. T. Choy ,  Prajakt V. Kulkarni ,  Samir Kumar Gupta

发明人： Song Wang ,  Eddie L. T. Choy ,  Prajakt V. Kulkarni ,  Samir Kumar Gupta

IPC分类号： H03M7/00

CPC分类号： H03H17/0685 ,  H03H17/0294

摘要： Techniques are described for sampling rate conversion in the digital domain by up-sampling and down-sampling a digital signal according to a selected intermediate sampling frequency. A prototype anti-aliasing filter that has a bandwidth with multiple factors is stored in memory. The techniques include selecting an intermediate sampling frequency to be an integer multiple of a desired output sampling frequency of a digital signal based on the factors of the prototype filter, and selecting a down-sampling factor to be the same integer associated with the selected intermediate sampling frequency. A filter generator generates an anti-aliasing filter for the selected down-sampling factor based on the prototype filter. A sampling rate converter up-samples the digital signal at an input sampling frequency to the selected intermediate sampling frequency, filters the digital signal with the derived anti-aliasing filter, and down-samples the digital signal by the selected down-sampling factor to the desired output sampling frequency.

摘要翻译： 描述了通过根据所选择的中间采样频率对数字信号进行上采样和下采样来对数字域中的采样率转换进行描述的技术。 具有多个因素的带宽的原型抗混叠滤波器存储在存储器中。 这些技术包括基于原型滤波器的因素来选择中间采样频率为数字信号的期望输出采样频率的整数倍，并且将下采样因子选择为与所选择的中间采样相关联的整数 频率。 滤波器发生器基于原型滤波器生成用于所选择的下采样因子的抗混叠滤波器。 采样率转换器将数字信号以输入采样频率向采样频率进行上采样，以采样导出的抗混叠滤波器对数字信号进行滤波，并通过选择的下采样因子将数字信号下采样到 所需输出采样频率。

公开(公告)号：US20090089053A1

公开(公告)日：2009-04-02

申请号：US11864897

申请日：2007-09-28

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

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

IPC分类号： G10L15/02 ,  G10L11/02

CPC分类号： G10L25/78 ,  G10L2021/02165

摘要： Voice activity detection using multiple microphones can be based on a relationship between an energy at each of a speech reference microphone and a noise reference microphone. The energy output from each of the speech reference microphone and the noise reference microphone can be determined. A speech to noise energy ratio can be determined and compared to a predetermined voice activity threshold. In another embodiment, the absolute value of the autocorrelation of the speech and noise reference signals are determined and a ratio based on autocorrelation values is determined. Ratios that exceed the predetermined threshold can indicate the presence of a voice signal. The speech and noise energies or autocorrelations can be determined using a weighted average or over a discrete frame size.

摘要翻译： 使用多个麦克风的语音活动检测可以基于语音基准麦克风和噪声参考麦克风各自的能量之间的关系。 可以确定来自每个语音参考麦克风和噪声参考麦克风的能量输出。 可以确定语音能量比，并将其与预定的语音活动阈值进行比较。 在另一个实施例中，确定语音和噪声参考信号的自相关的绝对值，并且确定基于自相关值的比率。 超过预定阈值的比率可以指示语音信号的存在。 可以使用加权平均值或离散的帧大小来确定语音和噪声能量或自相关性。

公开(公告)号：US07508327B2

公开(公告)日：2009-03-24

申请号：US11558313

申请日：2006-11-09

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

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

IPC分类号： H03M7/00

CPC分类号： H03H17/0685

摘要： In general, this disclosure describes techniques for changing a sampling frequency of a digital signal. In particular, the techniques provide a more accurate way to determining a relative timing between a desired output sample and a corresponding input sample using a non-approximated integer representation of the relative timing. The relative timing between the desired output sample and corresponding input sample may be represented using a first component that identifies a latest input sample of the digital signal used to generate intermediate samples, a second component that identifies an intermediate sample, and a third component that identifies a timing difference between the desired output sample and the intermediate sample. Each of the components may be recursively updated using non-approximated integer values.

摘要翻译： 通常，本公开描述了用于改变数字信号的采样频率的技术。 特别地，这些技术提供了使用相对定时的非近似整数表示来确定期望输出采样和相应输入采样之间的相对定时的更精确的方法。 可以使用标识用于生成中间样本的数字信号的最新输入样本的第一组件，标识中间样本的第二组件和标识中间样本的第三组件来表示期望输出样本与相应输入样本之间的相对时序 所需输出样本和中间样本之间的时间差。 可以使用非近似的整数值递归地更新每个组件。

公开(公告)号：US07742746B2

公开(公告)日：2010-06-22

申请号：US11742476

申请日：2007-04-30

申请人： Pei Xiang ,  Song Wang ,  Prajakt V. Kulkarni ,  Samir Kumar Gupta ,  Eddie L. T. Choy

发明人： Pei Xiang ,  Song Wang ,  Prajakt V. Kulkarni ,  Samir Kumar Gupta ,  Eddie L. T. Choy

IPC分类号： H04B1/00

CPC分类号： H03G7/007 ,  H03G3/32 ,  H04M1/6016

摘要： A mobile audio device (for example, a cellular telephone, personal digital audio player, or MP3 player) performs Audio Dynamic Range Control (ADRC) and Automatic Volume Control (AVC) to increase the volume of sound emitted from a speaker of the mobile audio device so that faint passages of the audio will be more audible. This amplification of faint passages occurs without overly amplifying other louder passages, and without substantial distortion due to clipping. Multi-Microphone Active Noise Cancellation (MMANC) functionality is, for example, used to remove background noise from audio information picked up on microphones of the mobile audio device. The noise-canceled audio may then be communicated from the device. The MMANC functionality generates a noise reference signal as an intermediate signal. The intermediate signal is conditioned and then used as a reference by the AVC process. The gain applied during the AVC process is a function of the noise reference signal.

摘要翻译： 移动音频设备（例如，蜂窝电话，个人数字音频播放器或MP3播放器）执行音频动态范围控制（ADRC）和自动音量控制（AVC）以增加从移动音频的扬声器发出的声音的音量 设备使得音频的微弱通道更可听见。 这种微弱通道的放大发生，而不会过度放大其他更大的通道，并且没有由于限幅导致的实质性变形。 例如，多麦克风有源噪声消除（MMANC）功能用于从移动音频设备的麦克风拾取的音频信息中去除背景噪声。 然后可以从设备传送噪声消除的音频。 MMANC功能产生噪声参考信号作为中间信号。 中间信号被调节，然后用作AVC处理的参考。 在AVC过程中应用的增益是噪声参考信号的函数。