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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

US20070286426A1 Mixing techniques for mixing audio 有权
标题翻译：混音技术混音
公开(公告)号：US20070286426A1
公开(公告)日：2007-12-13
申请号：US11449454
申请日：2006-06-07
申请人： Pei Xiang , Eddie L.T. Choy , Prajakt V. Kulkarni , Samir Kumar Gupta
发明人： Pei Xiang , Eddie L.T. Choy , Prajakt V. Kulkarni , Samir Kumar Gupta
IPC分类号： H04R5/00
CPC分类号： H04S7/30 , H04M1/6016 , H04M1/72558 , H04R2499/11
摘要： This disclosure describes audio mixing techniques that intelligently combine two or more audio signals into an output signal. The techniques allow audio to be combined, yet create perceptual differentiation between the different audio signals. The result is that a user is able to hear both audio signals in a combined output, but the different audio signals do not perceptually interfere with one another. The techniques are relatively simple to implement and are well suited for radio telephones.
摘要翻译：本公开描述了将两个或多个音频信号智能地组合成输出信号的音频混合技术。这些技术允许音频被组合，但是在不同音频信号之间产生感知差异。结果是用户能够听到组合输出中的两个音频信号，但是不同的音频信号不会感知到彼此干扰。这些技术实现起来相对简单，并且非常适合于无线电话。

4. 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。

5. 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。

6. 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。

7. 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。

8. 发明申请

US20080074542A1 METHOD AND SYSTEM FOR ERROR ROBUST AUDIO PLAYBACK TIME STAMP REPORTING 有权
标题翻译：用于错误的音频播放时间戳报告的方法和系统
公开(公告)号：US20080074542A1
公开(公告)日：2008-03-27
申请号：US11535406
申请日：2006-09-26
申请人： Mingxia Cheng , Anthony Patrick Mauro , Eddie L.T. Choy , Yujie Gao , Kuntal Dilipsinh Sampat , Matthew Blaine Zivney , Satish Goverdhan , Samir Kumar Gupta , Harinath Garudadri
发明人： Mingxia Cheng , Anthony Patrick Mauro , Eddie L.T. Choy , Yujie Gao , Kuntal Dilipsinh Sampat , Matthew Blaine Zivney , Satish Goverdhan , Samir Kumar Gupta , Harinath Garudadri
IPC分类号： H04N5/04
CPC分类号： H04N21/2368 , G10L19/005 , H04N19/436 , H04N19/68 , H04N19/89 , H04N21/4307 , H04N21/4341 , H04N21/439 , H04N21/44204 , H04N21/4425 , H04N21/8106
摘要： A method and system for resynchronizing an embedded multimedia system using bytes consumed in an audio decoder. The bytes consumed provides a mechanism to compensate for bit error handling and correction in a system that does not require re-transmission. The audio decoder keeps track of the bytes consumed and periodically reports the bytes consumed. A host microprocessor indexes the actual bytes consumed since bit errors may have been handled or corrected to a predetermined byte count to determine whether resynchronization is necessary.
摘要翻译：一种使用音频解码器中消耗的字节重新同步嵌入式多媒体系统的方法和系统。消耗的字节提供了一种补偿不需要重新传输的系统中的位错误处理和校正的机制。音频解码器跟踪消耗的字节，并定期报告消耗的字节。主机微处理器索引所消耗的实际字节，因为位错误可能被处理或校正到预定的字节数以确定是否需要重新同步。

9. 发明申请

US20090024397A1 UNIFIED FILTER BANK FOR PERFORMING SIGNAL CONVERSIONS 有权
标题翻译：用于执行信号转换的统一过滤器银行
公开(公告)号：US20090024397A1
公开(公告)日：2009-01-22
申请号：US12174498
申请日：2008-07-16
申请人： Sang-Uk Ryu , Eddie L.T. Choy , Nidish Ramachandra Kamath , Samir Kumar Gupta , Suresh Devalapalli
发明人： Sang-Uk Ryu , Eddie L.T. Choy , Nidish Ramachandra Kamath , Samir Kumar Gupta , Suresh Devalapalli
IPC分类号： G10L19/00
CPC分类号： G10L19/16 , G10L19/0212
摘要： A unified filter bank for performing signal conversions may include an interface that receives signal conversion commands in relation to multiple types of compressed audio bitstreams. The unified filter bank may also include a reconfigurable transform component that performs a transform as part of signal conversion for the multiple types of compressed audio bitstreams. The unified filter bank may also include complementary modules that perform complementary processing as part of the signal conversion for the multiple types of compressed audio bitstreams. The unified filter bank may also include an interface command controller that controls the configuration of the reconfigurable transform component and the complementary modules.
摘要翻译：用于执行信号转换的统一滤波器组可以包括接收关于多种类型的压缩音频位流的信号转换命令的接口。统一的滤波器组还可以包括可重新配置的变换组件，其执行作为多种类型的压缩音频位流的信号转换的一部分的变换。统一的滤波器组还可以包括作为多种类型的压缩音频比特流的信号转换的一部分执行补充处理的互补模块。统一的滤波器组还可以包括控制可重构变换组件和互补模块的配置的接口命令控制器。

10. 发明申请

US20080229919A1 AUDIO PROCESSING HARDWARE ELEMENTS 失效
标题翻译：音频处理硬件元件
公开(公告)号：US20080229919A1
公开(公告)日：2008-09-25
申请号：US12042181
申请日：2008-03-04
申请人： Nidish Ramachandra Kamath , Eddie L.T. Choy , Prajakt Kulkarni , Samir Kumar Gupta , Stephen Molloy , Suresh Devalapalli
发明人： Nidish Ramachandra Kamath , Eddie L.T. Choy , Prajakt Kulkarni , Samir Kumar Gupta , Stephen Molloy , Suresh Devalapalli
IPC分类号： G10H7/00
CPC分类号： G10H7/004 , G10H1/0066 , G10H2230/031 , G10H2250/541
摘要： This disclosure describes techniques that make use of a plurality of hardware elements that operate simultaneously to service synthesis parameters generated from one or more audio files, such as musical instrument digital interface (MIDI) files. In one example, a method comprises storing audio synthesis parameters generated for one or more audio files of an audio frame, processing a first audio synthesis parameter using a first audio processing element of a hardware unit to generate first audio information, processing a second audio synthesis parameter using a second audio processing element of the hardware unit to generate second audio information, and generating audio samples for the audio frame based at least in part on a combination of the first and second audio information.
摘要翻译：本公开描述了利用多个硬件元件的技术，该多个硬件元件同时工作来服务从诸如乐器数字接口（MIDI）文件的一个或多个音频文件生成的合成参数。在一个示例中，一种方法包括存储为音频帧的一个或多个音频文件生成的音频合成参数，使用硬件单元的第一音频处理元件处理第一音频合成参数以产生第一音频信息，处理第二音频合成参数，使用所述硬件单元的第二音频处理元件来产生第二音频信息，以及至少部分地基于所述第一和第二音频信息的组合来生成所述音频帧的音频样本。

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