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1. 发明授权

US09083994B2 Method and system for error robust audio playback time stamp reporting 有权
标题翻译：错误强大的音频播放时间戳报告的方法和系统
公开(公告)号：US09083994B2
公开(公告)日：2015-07-14
申请号：US11535406
申请日：2006-09-26
申请人： Mingxia Cheng , Anthony Patrick Mauro, II , 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, II , Eddie L. T. Choy , Yujie Gao , Kuntal Dilipsinh Sampat , Matthew Blaine Zivney , Satish Goverdhan , Samir Kumar Gupta , Harinath Garudadri
IPC分类号： G06F17/00 , H04N21/2368 , G10L19/005 , H04N21/43 , H04N21/434 , H04N21/439 , H04N21/442 , H04N21/4425 , H04N21/81 , H04N19/89 , H04N19/436 , H04N19/68
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。

2. 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。

3. 发明授权

US08532714B2 Dynamically provisioning a device with audio processing capability 有权
标题翻译：动态配置具有音频处理能力的设备
公开(公告)号：US08532714B2
公开(公告)日：2013-09-10
申请号：US12362098
申请日：2009-01-29
申请人： Kuntal Dilipsinh Sampat , Samir Kumar Gupta , Eddie L. T. Choy , Joel Linsky
发明人： Kuntal Dilipsinh Sampat , Samir Kumar Gupta , Eddie L. T. Choy , Joel Linsky
IPC分类号： H04B1/00
CPC分类号： H04W8/245 , G06F9/44526
摘要： An executable is downloaded to an audio output device over a communications link. The executable may configure the audio output device to decode audio encoded in a specified format. The executable may also or alternatively include other audio processing software. The audio may include voice and/or audio playback, e.g., music playback. The ability to download an audio executable allows dynamic provisioning of various decoding and/or audio process capabilities to an audio output device. This may eliminate the need to transcode digitized audio for playback at the audio output device, and may also allow the audio output device to decode multiple audio formats without having multiple audio decoders permanently residing within the audio output device.
摘要翻译：通过通信链路将可执行文件下载到音频输出设备。可执行程序可以配置音频输出设备来解码以指定格式编码的音频。可执行程序还可以或者可选地包括其他音频处理软件。音频可以包括语音和/或音频播放，例如音乐播放。下载音频可执行文件的能力允许向音频输出设备动态地提供各种解码和/或音频处理能力。这可以消除将数字化音频转码为在音频输出设备处回放的需要，并且还可以允许音频输出设备解码多种音频格式，而不会使多个音频解码器永久驻留在音频输出设备内。

4. 发明授权

US08725520B2 Power efficient batch-frame audio decoding apparatus, system and method 有权
标题翻译：高效率批量音频解码装置，系统及方法
公开(公告)号：US08725520B2
公开(公告)日：2014-05-13
申请号：US12204593
申请日：2008-09-04
申请人： Heejong Yoo , Nidish Ramachandra Kamath , Eddie L. T. Choy , Johnny K John , Samir Kumar Gupta
发明人： Heejong Yoo , Nidish Ramachandra Kamath , Eddie L. T. Choy , Johnny K John , Samir Kumar Gupta
IPC分类号： G10L19/00 , G06F17/00
CPC分类号： G10L19/16 , G06F1/3203 , G06F3/162 , H04W52/0274
摘要： Power savings in a mobile device is accomplished by generating audio samples by decoding a bitstream with a decoding system within the mobile device. The generated audio samples are transferred into at least one memory bank in a set of memory banks in a power saver block within the mobile device. Parts of the decoding system not involved in the storing of the generated audio samples are switched off after batch decoding a bitstream associated with multiple audio frames. The bitstream includes bits less than that found in one audio file. At least one of the memory banks in the set of memory banks is power collapsible. The fetching of the decoded by the decoding system can be synchronized with a paging channel of a modem in the mobile device. The transferred audio samples is a lossless compression and may occur after a re-encoding.
摘要翻译：移动设备中的功率节省通过利用移动设备内的解码系统对比特流进行解码来生成音频样本来实现。生成的音频样本被传送到移动设备内的节电块中的一组存储器组中的至少一个存储体。在对与多个音频帧相关联的比特流进行批量解码之后，不涉及生成的音频样本的存储的部分解码系统被关闭。比特流包括比在一个音频文件中发现的比特小的比特。存储器组中的至少一个存储体是电源可折叠的。由解码系统解码的提取可以与移动设备中的调制解调器的寻呼信道同步。传输的音频样本是无损压缩，并且可能在重新编码之后发生。

5. 发明授权

US08660280B2 Methods and apparatus for providing a distinct perceptual location for an audio source within an audio mixture 有权
标题翻译：为音频混合中的音频源提供不同感知位置的方法和装置
公开(公告)号：US08660280B2
公开(公告)日：2014-02-25
申请号：US11946365
申请日：2007-11-28
申请人： Pei Xiang , Samir Kumar Gupta , Eddie L. T. Choy , Prajakt V. Kulkarni
发明人： Pei Xiang , Samir Kumar Gupta , Eddie L. T. Choy , Prajakt V. Kulkarni
IPC分类号： H04R5/02 , H04R5/00
CPC分类号： H04S7/30 , H04S7/302 , H04S2400/11
摘要： In accordance with a method for providing a distinct perceptual location for an audio source within an audio mixture, a foreground signal may be processed to provide a foreground perceptual angle for the foreground signal. The foreground signal may also be processed to provide a desired attenuation level for the foreground signal. A background signal may be processed to provide a background perceptual angle for the background signal. The background signal may also be processed to provide a desired attenuation level for the background signal. The foreground signal and the background signal may be combined into an output audio source.
摘要翻译：根据用于为音频混合中的音频源提供不同的感知位置的方法，可以处理前景信号以提供前景信号的前景感知角度。还可以处理前景信号以为前景信号提供期望的衰减电平。可以处理背景信号以为背景信号提供背景感知角度。背景信号也可以被处理以为背景信号提供期望的衰减水平。前景信号和背景信号可以组合成输出音频源。

6. 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。

7. 发明授权

US08175871B2 Apparatus and method of noise and echo reduction in multiple microphone audio systems 有权
标题翻译：多麦克风音频系统的噪声和回波减少的装置和方法
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。

8. 发明授权

US08041057B2 Mixing techniques for mixing audio 有权
标题翻译：混音技术混音
公开(公告)号：US08041057B2
公开(公告)日：2011-10-18
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： H04B1/00
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。

9. 发明申请

US20090135976A1 RESOLVING BUFFER UNDERFLOW/OVERFLOW IN A DIGITAL SYSTEM 失效
标题翻译：解决缓冲区在数字系统中的下流/溢出
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。

10. 发明授权

US08515106B2 Methods and apparatus for providing an interface to a processing engine that utilizes intelligent audio mixing techniques 有权
标题翻译：用于提供利用智能音频混合技术的处理引擎的接口的方法和装置
公开(公告)号：US08515106B2
公开(公告)日：2013-08-20
申请号：US11946319
申请日：2007-11-28
申请人： Pei Xiang , Samir Kumar Gupta , Eddie L. T. Choy , Prajakt V. Kulkarni
发明人： Pei Xiang , Samir Kumar Gupta , Eddie L. T. Choy , Prajakt V. Kulkarni
IPC分类号： H04R5/00 , H04R5/02
CPC分类号： H04S7/30 , H04R2499/11 , H04S7/302 , H04S2400/11
摘要： A method for providing an interface to a processing engine that utilizes intelligent audio mixing techniques may include receiving a request to change a perceptual location of an audio source within an audio mixture from a current perceptual location relative to a listener to a new perceptual location relative to the listener. The audio mixture may include at least two audio sources. The method may also include generating one or more control signals that are configured to cause the processing engine to change the perceptual location of the audio source from the current perceptual location to the new perceptual location via separate foreground processing and background processing. The method may also include providing the one or more control signals to the processing engine.
摘要翻译：用于向使用智能音频混合技术的处理引擎提供接口的方法可以包括接收将音频混合内的音频源的感知位置从相对于收听者的当前感知位置改变为相对于听觉位置的新感知位置的请求听众。音频混合可以包括至少两个音频源。该方法还可以包括生成一个或多个控制信号，该控制信号被配置为使得处理引擎通过分离的前景处理和后台处理将音频源的感知位置从当前的感知位置改变到新的感知位置。该方法还可以包括向处理引擎提供一个或多个控制信号。

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