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1. 发明申请

US20120215546A1 Complexity Scalable Perceptual Tempo Estimation 有权
标题翻译：复杂性可扩展感知速度估计
公开(公告)号：US20120215546A1
公开(公告)日：2012-08-23
申请号：US13503136
申请日：2010-10-26
申请人： Arijit Biswas , Danilo Hollosi , Michael Schug
发明人： Arijit Biswas , Danilo Hollosi , Michael Schug
IPC分类号： G10L19/00
CPC分类号： G10H1/40 , G10H2210/076 , G10H2230/015 , G10H2240/075
摘要： The present document relates to methods and systems for estimating the tempo of a media signal, such as audio or combined video/audio signal. In particular, the document relates to the estimation of tempo perceived by human listeners, as well as to methods and systems for tempo estimation at scalable computational complexity. A method and system for extracting tempo information of an audio signal from an encoded bit-stream of the audio signal comprising spectral band replication data is described. The method comprises the steps of determining a payload quantity associated with the amount of spectral band replication data comprised in the encoded bit-stream for a time interval of the audio signal; repeating the determining step for successive time intervals of the encoded bit-stream of the audio signal, thereby determining a sequence of payload quantities; identifying a periodicity in the sequence of payload quantities; and extracting tempo information of the audio signal from the identified periodicity.
摘要翻译：本文件涉及用于估计媒体信号（诸如音频或组合视频/音频信号）的速度的方法和系统。特别地，该文件涉及人类听众感知的节奏的估计，以及用于以可缩放的计算复杂度进行速度估计的方法和系统。描述用于从包括频谱带复制数据的音频信号的编码比特流中提取音频信号的速度信息的方法和系统。该方法包括以下步骤：在音频信号的时间间隔中确定与包含在编码比特流中的频谱带复制数据量相关联的有效载荷数量; 重复音频信号的编码比特流的连续时间间隔的确定步骤，从而确定有效载荷量的序列; 识别有效载荷数量序列中的周期; 以及从所识别的周期中提取音频信号的速度信息。

2. 发明授权

US09466275B2 Complexity scalable perceptual tempo estimation 有权
标题翻译：复杂性可扩展感知速度估计
公开(公告)号：US09466275B2
公开(公告)日：2016-10-11
申请号：US13503136
申请日：2010-10-26
申请人： Arijit Biswas , Danilo Hollosi , Michael Schug
发明人： Arijit Biswas , Danilo Hollosi , Michael Schug
IPC分类号： G10H7/00 , G10H1/40
CPC分类号： G10H1/40 , G10H2210/076 , G10H2230/015 , G10H2240/075
摘要： The present document relates to methods and systems for estimating the tempo of a media signal, such as audio or combined video/audio signal. In particular, the document relates to the estimation of tempo perceived by human listeners, as well as to methods and systems for tempo estimation at scalable computational complexity. A method and system for extracting tempo information of an audio signal from an encoded bit-stream of the audio signal comprising spectral band replication data is described. The method comprises the steps of determining a payload quantity associated with the amount of spectral band replication data comprised in the encoded bit-stream for a time interval of the audio signal; repeating the determining step for successive time intervals of the encoded bit-stream of the audio signal, thereby determining a sequence of payload quantities; identifying a periodicity in the sequence of payload quantities; and extracting tempo information of the audio signal from the identified periodicity.
摘要翻译：本文件涉及用于估计媒体信号（诸如音频或组合视频/音频信号）的速度的方法和系统。特别地，该文件涉及人类听众感知的节奏的估计，以及用于以可缩放的计算复杂度进行速度估计的方法和系统。描述用于从包括频谱带复制数据的音频信号的编码比特流中提取音频信号的速度信息的方法和系统。该方法包括以下步骤：在音频信号的时间间隔中确定与包含在编码比特流中的频谱带复制数据量相关联的有效载荷数量; 重复音频信号的编码比特流的连续时间间隔的确定步骤，从而确定有效载荷量的序列; 识别有效载荷数量序列中的周期; 以及从所识别的周期中提取音频信号的速度信息。

3. 发明申请

US20130179175A1 Method and System for Encoding Audio Data with Adaptive Low Frequency Compensation 有权
标题翻译：用自适应低频补偿编码音频数据的方法和系统
公开(公告)号：US20130179175A1
公开(公告)日：2013-07-11
申请号：US13588890
申请日：2012-08-17
申请人： Arijit Biswas , Vinay Melkote , Michael Schug , Grant A. Davidson , Mark S. Vinton
发明人： Arijit Biswas , Vinay Melkote , Michael Schug , Grant A. Davidson , Mark S. Vinton
IPC分类号： G10L19/00
CPC分类号： G10L19/028 , G10L19/0204 , G10L19/032 , G10L19/265
摘要： A method for determining mantissa bit allocation of frequency domain audio data to be encoded, including by performing adaptive low frequency compensation on each frequency band of a set of low frequency bands of the data. The low frequency compensation includes steps of: performing tonality detection on the audio data to generate compensation control data indicative of whether each frequency band in the set has prominent tonal content; and performing low frequency compensation on each frequency band in the set having prominent tonal content, including by correcting a preliminary masking value for each frequency band having prominent tonal content, but not performing low frequency compensation on the audio data in any other frequency band in the set. Other aspects are audio encoding methods including such tonality detection and low frequency compensation steps, and a system configured to perform any embodiment of the inventive method.
摘要翻译：一种用于确定要编码的频域音频数据的尾数位分配的方法，包括通过对数据的一组低频带的每个频带执行自适应低频补偿。低频补偿包括以下步骤：对音频数据执行音调检测，以产生指示该组中的每个频带是否具有突出的音调内容的补偿控制数据; 并且对具有突出音调内容的集合中的每个频带执行低频补偿，包括通过校正具有突出音调内容的每个频带的初步屏蔽值，但不对低频补偿中的任何其他频带中的音频数据执行低频补偿组。其他方面是包括这种音调检测和低频补偿步骤的音频编码方法，以及被配置为执行本发明方法的任何实施例的系统。

4. 发明授权

US08527264B2 Method and system for encoding audio data with adaptive low frequency compensation 有权
标题翻译：用自适应低频补偿编码音频数据的方法和系统
公开(公告)号：US08527264B2
公开(公告)日：2013-09-03
申请号：US13588890
申请日：2012-08-17
申请人： Arijit Biswas , Vinay Melkote , Michael Schug , Grant Allen Davidson , Mark Stuart Vinton
发明人： Arijit Biswas , Vinay Melkote , Michael Schug , Grant Allen Davidson , Mark Stuart Vinton
IPC分类号： G10L19/00
CPC分类号： G10L19/028 , G10L19/0204 , G10L19/032 , G10L19/265
摘要： A method for determining mantissa bit allocation of frequency domain audio data to be encoded, including by performing adaptive low frequency compensation on each frequency band of a set of low frequency bands of the data. The low frequency compensation includes steps of: performing tonality detection on the audio data to generate compensation control data indicative of whether each frequency band in the set has prominent tonal content; and performing low frequency compensation on each frequency band in the set having prominent tonal content, including by correcting a preliminary masking value for each frequency band having prominent tonal content, but not performing low frequency compensation on the audio data in any other frequency band in the set; wherein the frequency domain audio data comprises an exponent value for said each low frequency band of the set, and the tonality detection includes determining, for said each low frequency band of the set, a measure of difference between exponents and corresponding tented exponents of the audio data. Other aspects are audio encoding methods including such tonality detection and low frequency compensation steps, and a system configured to perform any embodiment of the inventive method.
摘要翻译：一种用于确定要编码的频域音频数据的尾数位分配的方法，包括通过对数据的一组低频带的每个频带执行自适应低频补偿。低频补偿包括以下步骤：对音频数据执行音调检测，以产生指示该组中的每个频带是否具有突出的音调内容的补偿控制数据; 并且对具有突出音调内容的集合中的每个频带执行低频补偿，包括通过校正具有突出音调内容的每个频带的初步屏蔽值，但不对低频补偿中的任何其他频带中的音频数据执行低频补偿组; 其中所述频域音频数据包括所述组的所述每个低频带的指数值，并且所述音调检测包括针对所述组的所述每个低频带确定所述音频的指数和对应的帐篷指数之间的差的度量数据。其他方面是包括这种音调检测和低频补偿步骤的音频编码方法，以及被配置为执行本发明方法的任何实施例的系统。

5. 发明申请

US20130289756A1 Ranking Representative Segments in Media Data 有权
标题翻译：媒体数据中的代表片段排名
公开(公告)号：US20130289756A1
公开(公告)日：2013-10-31
申请号：US13997866
申请日：2011-12-15
申请人： Barbara Resch , Regunathan Radhakrishnan , Arijit Biswas , Jonas Engdegard
发明人： Barbara Resch , Regunathan Radhakrishnan , Arijit Biswas , Jonas Engdegard
IPC分类号： G06F17/00
CPC分类号： H04R29/00 , G06F17/00 , G06F17/3053 , G10H1/0008 , G10H2210/061 , G10H2240/151 , G10L25/48
摘要： Techniques for ranking representative segments in media data are provided. Media features of many different types may be extracted from the media data. A plurality of ranking scores may be assigned to a plurality of candidate representative segments. Each individual candidate representative segment in the plurality of candidate representative segments comprises at least one scene in one or more statistical patterns in media features of the media data based on one or more types of features extractable from the media data. Each individual ranking score in the plurality of ranking scores may be assigned to an individual candidate representative segment in the plurality of candidate representative segments. A representative segment to be played to an end user may be selected from the candidate representative segments, based on the plurality of ranking scores.
摘要翻译：提供了在媒体数据中排列代表片段的技术。可以从媒体数据中提取许多不同类型的媒体特征。可以将多个排名得分分配给多个候选代表段。基于从媒体数据可提取的一种或多种类型的特征，多个候选代表段中的每个候选代表段包括媒体数据的媒体特征中的一个或多个统计模式中的至少一个场景。可以将多个排名得分中的每个个体排名分数分配给多个候选代表段中的个人候选代表段。可以基于多个排名得分从候选代表段中选择要向最终用户播放的代表段。

6. 发明申请

US20110161630A1 GENERAL PURPOSE HARDWARE TO REPLACE FAULTY CORE COMPONENTS THAT MAY ALSO PROVIDE ADDITIONAL PROCESSOR FUNCTIONALITY 有权
标题翻译：一般用途五金替代可能还会提供附加处理器功能的故障核心组件
公开(公告)号：US20110161630A1
公开(公告)日：2011-06-30
申请号：US12647940
申请日：2009-12-28
申请人： Steven E. Raasch , Michael D. Powell , Shubhendu S. Mukherjee , Arijit Biswas
发明人： Steven E. Raasch , Michael D. Powell , Shubhendu S. Mukherjee , Arijit Biswas
IPC分类号： G06F11/07 , G06F12/08 , G06F9/30
CPC分类号： G06F9/3885 , G06F9/30196 , G06F9/3857 , G06F9/3861 , G06F9/3867 , G06F9/3877 , G06F11/2028 , G06F11/2038 , G06F11/2043
摘要： An apparatus and method is described herein for replacing faulty core components. General purpose hardware is provided to replace core pipeline components, such as execution units. In the embodiment of execution unit replacement, a proxy unit is provided, such that mapping logic is able to map instruction/operations, which correspond to faulty execution units, to the proxy unit. As a result, the proxy unit is able to receive the operations, send them to general purpose hardware for execution, and subsequently write-back the execution results to a register file; it essentially replaces the defective execution unit allowing a processor with defective units to be sold or continue operation.
摘要翻译：本文描述了用于替换故障核心部件的装置和方法。提供通用硬件来代替核心管道组件，如执行单元。在执行单元替换的实施例中，提供代理单元，使得映射逻辑能够将与故障执行单元相对应的指令/操作映射到代理单元。因此，代理单元能够接收操作，将其发送到通用硬件执行，然后将执行结果写回寄存器文件; 它基本上取代了有缺陷的执行单元，允许出售有缺陷单元的处理器或继续操作。

7. 发明授权

US07849387B2 Detecting architectural vulnerability of processor resources 有权
标题翻译：检测处理器资源的架构漏洞
公开(公告)号：US07849387B2
公开(公告)日：2010-12-07
申请号：US12148812
申请日：2008-04-23
申请人： Arijit Biswas , Niranjan Soundararajan , Shubhendu Mukherjee
发明人： Arijit Biswas , Niranjan Soundararajan , Shubhendu Mukherjee
IPC分类号： H03M13/00
CPC分类号： G06F11/3466 , G06F11/3409 , G06F2201/88 , Y02D10/34
摘要： In one embodiment, a quantum detector is provided to detect a vulnerability measure for a processor based on a processor metrics each associated with operation of a processor structure during a quantum, along with a controller to control an error mitigation unit based on the vulnerability measure. Other embodiments are described and claimed.
摘要翻译：在一个实施例中，提供量子检测器，用于基于与量化期间的处理器结构的操作相关联的处理器度量，以及基于所述漏洞测量来控制误差减轻单元的控制器，来检测处理器的脆弱性度量。描述和要求保护其他实施例。

8. 发明授权

US07747932B2 Reducing the uncorrectable error rate in a lockstepped dual-modular redundancy system 失效
标题翻译：在锁定的双模块冗余系统中减少不可纠正的错误率
公开(公告)号：US07747932B2
公开(公告)日：2010-06-29
申请号：US11173835
申请日：2005-06-30
申请人： Paul B. Racunas , Joel S. Emer , Arijit Biswas , Shubhendu S. Mukherjee , Steven E. Raasch
发明人： Paul B. Racunas , Joel S. Emer , Arijit Biswas , Shubhendu S. Mukherjee , Steven E. Raasch
IPC分类号： H03M13/00
CPC分类号： G06F11/165 , G06F11/1641 , G06F11/1658 , G06F11/1683
摘要： Embodiments of apparatuses and methods for reducing the uncorrectable error rate in a lockstepped dual-modular redundancy system are disclosed. In one embodiment, an apparatus includes two processor cores, a micro-checker, a global checker, and fault logic. The micro-checker is to detect whether a value from a structure in one core matches a value from the corresponding structure in the other core. The global checker is to detect lockstep failures between the two cores. The fault logic is to cause the two cores to be resynchronized if there is a lockstep error but the micro-checker has detected a mismatch.
摘要翻译：公开了用于降低锁定双模块冗余系统中的不可校正错误率的装置和方法的实施例。在一个实施例中，装置包括两个处理器核，微检查器，全局检验器和故障逻辑。微检查器是检测来自一个核心中的结构的值是否与另一个核心中的相应结构的值匹配。全局检查器是检测两个内核之间的锁步失败。如果存在锁步错误但微检查器检测到不匹配，则故障逻辑是使两个内核重新同步。

9. 发明授权

US09081688B2 Obtaining data for redundant multithreading (RMT) execution 有权
标题翻译：获取冗余多线程（RMT）执行的数据
公开(公告)号：US09081688B2
公开(公告)日：2015-07-14
申请号：US12317801
申请日：2008-12-30
申请人： Glenn J. Hinton , Steven E. Raasch , Sebastien Hily , John G. Holm , Ronak Singhal , Avinash Sodani , Deborah T. Marr , Shubhendu S. Mukherjee , Arijit Biswas , Adrian C. Moga
发明人： Glenn J. Hinton , Steven E. Raasch , Sebastien Hily , John G. Holm , Ronak Singhal , Avinash Sodani , Deborah T. Marr , Shubhendu S. Mukherjee , Arijit Biswas , Adrian C. Moga
IPC分类号： G06F12/08
CPC分类号： G06F12/0842 , G06F12/0817
摘要： In one embodiment, the present invention includes a method for providing a cache block in an exclusive state to a first cache and providing the same cache block in the exclusive state to a second cache when cores accessing the two caches are executing redundant threads. Other embodiments are described and claimed.
摘要翻译：在一个实施例中，本发明包括一种用于向第一高速缓存提供处于独占状态的高速缓存块的方法，并且当访问两个高速缓存的核心正在执行冗余线程时，以独占状态将相同的高速缓存块提供给第二高速缓存。描述和要求保护其他实施例。

10. 发明授权

US08494863B2 Audio encoder and decoder with long term prediction 有权
标题翻译：音频编码器和解码器具有长期预测
公开(公告)号：US08494863B2
公开(公告)日：2013-07-23
申请号：US12811419
申请日：2008-12-30
申请人： Arijit Biswas , Heiko Purnhagen , Kristofer Kjoerling , Barbara Resch , Lars Villemoes , Per Hedelin
发明人： Arijit Biswas , Heiko Purnhagen , Kristofer Kjoerling , Barbara Resch , Lars Villemoes , Per Hedelin
IPC分类号： G10I19/14 , G10L19/06 , G10L19/00 , G10L19/12 , G10L21/04
CPC分类号： G10L19/26 , G10L19/008 , G10L19/032 , G10L19/035
摘要： The present invention teaches a new audio coding system that can code both general audio and speech signals well at low bit rates. A proposed audio coding system comprises a linear prediction unit for filtering an input signal based on an adaptive filter; a transformation unit for transforming a frame of the filtered input signal into a transform domain; a quantization unit for quantizing a transform domain signal; a long term prediction unit for determining an estimation of the frame of the filtered input signal based on a reconstruction of a previous segment of the filtered input signal; and a transform domain signal combination unit for combining, in the transform domain, the long term prediction estimation and the transformed input signal to generate the transform domain signal.
摘要翻译：本发明教导了一种新的音频编码系统，其可以以低比特率良好地对一般音频和语音信号进行编码。所提出的音频编码系统包括：基于自适应滤波器对输入信号进行滤波的线性预测单元; 变换单元，用于将经滤波的输入信号的帧变换为变换域; 用于量化变换域信号的量化单元; 一个长期预测单元，用于基于经滤波的输入信号的先前段的重建来确定滤波输入信号的帧的估计; 以及变换域信号组合单元，用于在变换域中组合长期预测估计和变换的输入信号，以生成变换域信号。

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