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1. 发明申请

US20070192390A1 Digital domain sampling rate converter 有权
标题翻译：数字域采样率转换器
公开(公告)号：US20070192390A1
公开(公告)日：2007-08-16
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： G06F1/02
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。

2. 发明申请

US20080269926A1 AUTOMATIC VOLUME AND DYNAMIC RANGE ADJUSTMENT FOR MOBILE AUDIO DEVICES 有权
标题翻译：自动音量和动态范围调整移动音频设备
公开(公告)号：US20080269926A1
公开(公告)日：2008-10-30
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： H04S7/00
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。

3. 发明申请

US20070290900A1 INTEGER REPRESENATION OF RELATIVE TIMING BETWEEN DESIRED OUTPUT SAMPLES AND CORRESPONDING INPUT SAMPLES 有权
标题翻译：所有输出样本和相应输入样本之间相对时间的整体表示
公开(公告)号：US20070290900A1
公开(公告)日：2007-12-20
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。

4. 发明申请

US20070257840A1 ENHANCEMENT TECHNIQUES FOR BLIND SOURCE SEPARATION (BSS) 有权
标题翻译：盲源分离（BSS）的增强技术
公开(公告)号：US20070257840A1
公开(公告)日：2007-11-08
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： H04J11/00
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。

5. 发明授权

US07528745B2 Digital domain sampling rate converter 有权
标题翻译：数字域采样率转换器
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。

6. 发明授权

US07742746B2 Automatic volume and dynamic range adjustment for mobile audio devices 有权
标题翻译：移动音频设备的自动音量和动态范围调整
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。

7. 发明申请

US20090196429A1 SIGNALING MICROPHONE COVERING TO THE USER 有权
标题翻译：信号麦克风覆盖用户
公开(公告)号：US20090196429A1
公开(公告)日：2009-08-06
申请号：US12023970
申请日：2008-01-31
申请人： Dinesh Ramakrishnan , Ravi Satyanarayanan , Song Wang , Eddie L.T. Choy
发明人： Dinesh Ramakrishnan , Ravi Satyanarayanan , Song Wang , Eddie L.T. Choy
IPC分类号： H04R5/00
CPC分类号： H04R3/005 , H04R29/004 , H04R29/006 , H04R2499/11
摘要： A mechanism is provided that monitors secondary microphone signals, in a multi-microphone mobile device, to warn the user if one or more secondary microphones are covered while the mobile device is in use. In one example, smoothly averaged power estimates of the secondary microphones may be computed and compared against the noise floor estimate of a primary microphone. Microphone covering detection may be made by comparing the secondary microphone smooth power estimates to the noise floor estimate for the primary microphone. In another example, the noise floor estimates for the primary and secondary microphone signals may be compared to the difference in the sensitivity of the first and second microphones to determine if the secondary microphone is covered. Once detection is made, a warning signal may be generated and issued to the user.
摘要翻译：提供了一种在多麦克风移动设备中监视次级麦克风信号的机制，以在移动设备正在使用时覆盖一个或多个辅助麦克风来警告用户。在一个示例中，可以计算二次麦克风的平滑平均功率估计，并将其与主麦克风的噪声基底估计进行比较。麦克风覆盖检测可以通过将次级麦克风平滑功率估计与主麦克风的噪声基底估计值进行比较来进行。在另一示例中，可以将主麦克风信号和次麦克风信号的噪声基底估计值与第一和第二麦克风的灵敏度差进行比较，以确定次麦克风是否被覆盖。一旦检测到，可能会产生一个警告信号并发给用户。

8. 发明申请

US20090190769A1 SOUND QUALITY BY INTELLIGENTLY SELECTING BETWEEN SIGNALS FROM A PLURALITY OF MICROPHONES 有权
标题翻译：通过智能选择来自多个麦克风的信号进行声音质量
公开(公告)号：US20090190769A1
公开(公告)日：2009-07-30
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： H04B3/20
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。分别从第一和第二麦克风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。

9. 发明申请

US20090190774A1 ENHANCED BLIND SOURCE SEPARATION ALGORITHM FOR HIGHLY CORRELATED MIXTURES 有权
标题翻译：用于高相关混合物的增强型盲源分离算法
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。

10. 发明授权

US08650238B2 Resolving buffer underflow/overflow in a digital system 失效
标题翻译：在数字系统中解决缓冲区下溢/溢出
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。

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