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1. 发明授权

US07663051B2 Audio processing hardware elements 失效
标题翻译：音频处理硬件元素
公开(公告)号：US07663051B2
公开(公告)日：2010-02-16
申请号：US12042181
申请日：2008-03-04
申请人： Nidish Kamath , Eddie L. T. Choy , Prajakt Kulkarni , Samir K Gupta , Stephen Molloy , Suresh Devalapalli
发明人： Nidish Kamath , Eddie L. T. Choy , Prajakt Kulkarni , Samir K Gupta , Stephen Molloy , Suresh Devalapalli
IPC分类号： G10H1/00
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。

2. 发明授权

US07663046B2 Pipeline techniques for processing musical instrument digital interface (MIDI) files 失效
标题翻译：用于处理乐器数字接口（MIDI）文件的管道技术
公开(公告)号：US07663046B2
公开(公告)日：2010-02-16
申请号：US12042170
申请日：2008-03-04
申请人： Prajakt Kulkarni , Eddie L. T. Choy , Nidish Ramachandra Kamath , Samir K Gupta , Stephen Molloy , Suresh Devalapalli
发明人： Prajakt Kulkarni , Eddie L. T. Choy , Nidish Ramachandra Kamath , Samir K Gupta , Stephen Molloy , Suresh Devalapalli
IPC分类号： G10H1/00
CPC分类号： G10H1/0066 , G10H7/004
摘要： This disclosure describes techniques for processing audio files that comply with the musical instrument digital interface (MIDI) format. In particular, various tasks associated with MIDI file processing are delegated between software operating on a general purpose processor, firmware associated with a digital signal processor (DSP), and dedicated hardware that is specifically designed for MIDI file processing. Alternatively, a multi-threaded DSP may be used instead of a general purpose processor and the DSP. In one aspect, this disclosure provides a method comprising parsing MIDI files and scheduling MIDI events associated with the MIDI files using a first process, processing the MIDI events using a second process to generate MIDI synthesis parameters, and generating audio samples using a hardware unit based on the synthesis parameters.
摘要翻译：本公开描述了用于处理符合乐器数字接口（MIDI）格式的音频文件的技术。具体而言，与在通用处理器上运行的软件，与数字信号处理器（DSP）相关联的固件以及专门为MIDI文件处理而专门设计的专用硬件之间的任何与MIDI文件处理相关的各种任务被委派。或者，可以使用多线程DSP来代替通用处理器和DSP。在一个方面，本公开提供了一种方法，包括使用第一处理解析MIDI文件和调度与MIDI文件相关联的MIDI事件，使用第二处理来处理MIDI事件以产生MIDI合成参数，以及使用基于硬件单元生成音频样本对合成参数。

3. 发明申请

US20080229918A1 PIPELINE TECHNIQUES FOR PROCESSING MUSICAL INSTRUMENT DIGITAL INTERFACE (MIDI) FILES 失效
标题翻译：用于处理音乐数字接口（MIDI）文件的管道技术
公开(公告)号：US20080229918A1
公开(公告)日：2008-09-25
申请号：US12042170
申请日：2008-03-04
申请人： Prajakt Kulkarni , Eddie L. T. Choy , Nidish Ramachandra Kamath , Samir Kumar Gupta , Stephen Molloy , Suresh Devalapalli
发明人： Prajakt Kulkarni , Eddie L. T. Choy , Nidish Ramachandra Kamath , Samir Kumar Gupta , Stephen Molloy , Suresh Devalapalli
IPC分类号： G10H7/00
CPC分类号： G10H1/0066 , G10H7/004
摘要： This disclosure describes techniques for processing audio files that comply with the musical instrument digital interface (MIDI) format. In particular, various tasks associated with MIDI file processing are delegated between software operating on a general purpose processor, firmware associated with a digital signal processor (DSP), and dedicated hardware that is specifically designed for MIDI file processing. Alternatively, a multi-threaded DSP may be used instead of a general purpose processor and the DSP. In one aspect, this disclosure provides a method comprising parsing MIDI files and scheduling MIDI events associated with the MIDI files using a first process, processing the MIDI events using a second process to generate MIDI synthesis parameters, and generating audio samples using a hardware unit based on the synthesis parameters.
摘要翻译：本公开描述了用于处理符合乐器数字接口（MIDI）格式的音频文件的技术。具体而言，与在通用处理器上运行的软件，与数字信号处理器（DSP）相关联的固件以及专门为MIDI文件处理而专门设计的专用硬件之间的任何与MIDI文件处理相关的各种任务被委派。或者，可以使用多线程DSP来代替通用处理器和DSP。在一个方面，本公开提供了一种方法，包括使用第一处理解析MIDI文件和调度与MIDI文件相关联的MIDI事件，使用第二处理来处理MIDI事件以产生MIDI合成参数，以及使用基于硬件单元生成音频样本对合成参数。

4. 发明授权

US08185381B2 Unified filter bank for performing signal conversions 有权
标题翻译：用于执行信号转换的统一滤波器组
公开(公告)号：US08185381B2
公开(公告)日：2012-05-22
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： G10L11/00 , G10L19/00 , G09G5/00 , H04H20/71 , H04L27/10 , H04J3/00 , H04K1/02
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。

5. 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。

6. 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。

7. 发明授权

US08615398B2 Audio coding selection based on device operating condition 有权
标题翻译：基于设备运行条件的音频编码选择
公开(公告)号：US08615398B2
公开(公告)日：2013-12-24
申请号：US12362119
申请日：2009-01-29
申请人： Kuntal Dilipsinh Sampat , Eddie L. T. Choy , Joel Linsky
发明人： Kuntal Dilipsinh Sampat , Eddie L. T. Choy , Joel Linsky
IPC分类号： G10L19/00
CPC分类号： H04R1/1083 , G10L19/22 , H03G3/32 , H04M1/6066 , H04R27/00 , H04R2227/003 , H04R2420/07 , H04R2460/03
摘要： A sensor is configured to determine at least one operating condition of a device and a selector is configured to select an audio coding process for the device, based on the operating condition. The operating condition may include remaining battery life of the device and/or ambient noise level. The selected audio coding process may consume less power than another possible audio coding process during audio processing. The audio may include voice and/or audio playback, e.g., music playback.
摘要翻译：传感器被配置为确定设备的至少一个操作条件，并且选择器被配置为基于操作条件为设备选择音频编码处理。操作条件可以包括设备的电池寿命和/或环境噪声水平。在音频处理期间，所选择的音频编码处理可能消耗比另一可能的音频编码处理更少的功率。音频可以包括语音和/或音频播放，例如音乐播放。

8. 发明授权

US08620644B2 Encoder-assisted frame loss concealment techniques for audio coding 有权
标题翻译：用于音频编码的编码器辅助帧丢失隐藏技术
公开(公告)号：US08620644B2
公开(公告)日：2013-12-31
申请号：US11431733
申请日：2006-05-10
申请人： Sang-Uk Ryu , Eddie L. T. Choy , Samir Kumar Gupta
发明人： Sang-Uk Ryu , Eddie L. T. Choy , Samir Kumar Gupta
IPC分类号： G10L19/00
CPC分类号： G10L19/005 , G10L19/02
摘要： Encoder-assisted frame loss concealment (FLC) techniques for decoding audio signals are described. A decoder may discard an erroneous frame of an audio signal and may implement the encoder-assisted FLC techniques in order to accurately conceal the discarded frame based on neighboring frames and side-information transmitted from the encoder. The encoder-assisted FLC techniques include estimating magnitudes of frequency-domain data for the frame based on frequency-domain data of neighboring frames, and estimating signs of the frequency-domain data based on a subset of signs transmitted from the encoder as side-information. Frequency-domain data for a frame of an audio signal includes tonal components and noise components. Signs estimated from a random signal may be substantially accurate for the noise components of the frequency-domain data. However, to achieve highly accurate sign estimation for the tonal components, the encoder transmits signs for the tonal components of the frequency-domain data as side-information.
摘要翻译：描述了用于解码音频信号的编码器辅助帧丢失隐藏（FLC）技术。解码器可以丢弃音频信号的错误帧，并且可以实现编码器辅助的FLC技术，以便基于相邻帧和从编码器发送的侧信息来准确地隐藏丢弃的帧。编码器辅助FLC技术包括基于相邻帧的频域数据估计帧的频域数据的幅度，并且基于从编码器发送的符号子集来估计频域数据的符号作为侧信息。音频信号的帧的频域数据包括音调分量和噪声分量。从随机信号估计的符号对于频域数据的噪声分量可能是基本准确的。然而，为了实现对音调分量的高度准确的符号估计，编码器将用于频域数据的音调分量的符号作为侧信息发送。

9. 发明授权

US08554551B2 Systems, methods, and apparatus for context replacement by audio level 有权
标题翻译：通过音频级别进行上下文替换的系统，方法和装置
公开(公告)号：US08554551B2
公开(公告)日：2013-10-08
申请号：US12129483
申请日：2008-05-29
申请人： Nagendra Nagaraja , Khaled El-Maleh , Eddie L. T. Choy
发明人： Nagendra Nagaraja , Khaled El-Maleh , Eddie L. T. Choy
IPC分类号： G10L21/00
CPC分类号： G10L21/02 , G10L19/012 , G10L21/0208 , G10L21/0272
摘要： Configurations disclosed herein include systems, methods, and apparatus that may be applied in a voice communications and/or storage application to remove, enhance, and/or replace the existing context. Enhancing the context of a voice communication may first include suppressing an existing context component from the digital audio signal to obtain a context suppressed signal. This signal may then be mixed with a new context signal to create a context enhanced signal, which may then be encoded before transmission. When this new context enhanced signal includes a speech component, it may be encoded and transmitted at a particular bit rate. When the context enhanced signal does not include a speech component, it may also be encoded at a similar bit rate. However, depending on the state of a process control signal, portions of a digital audio signal that lack a speech component may also be transmitted at a lower bit rate.
摘要翻译：本文公开的配置包括可以应用于语音通信和/或存储应用中以去除，增强和/或替代现有上下文的系统，方法和装置。增强语音通信的上下文可以首先包括从数字音频信号中抑制现有上下文成分以获得上下文抑制信号。然后可以将该信号与新的上下文信号混合以产生上下文增强信号，然后可以在传输之前进行编码。当该新的上下文增强信号包括一个语音分量时，它可以被编码并以特定比特率发送。当上下文增强信号不包括语音分量时，也可以以相似的比特率进行编码。然而，根据处理控制信号的状态，也可以以较低的比特率发送缺少语音分量的数字音频信号的部分。

10. 发明授权

US08411880B2 Sound quality by intelligently selecting between signals from a plurality of microphones 有权
标题翻译：通过智能地在多个麦克风的信号之间选择声音质量
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。

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