公开(公告)号：US06460153B1

公开(公告)日：2002-10-01

申请号：US09276955

申请日：1999-03-26

Applicant: Philip A. Chou ,  Sanjeev Mehrotra ,  Albert S. Wang

Inventor： Philip A. Chou ,  Sanjeev Mehrotra ,  Albert S. Wang

IPC: H03M1300

CPC classification number: H04N19/89

Abstract: A projection onto convex sets (POCS)-based method for consistent reconstruction of a signal from a subset of quantized coefficients received from an N×K overcomplete transform. By choosing a frame operator F to be the concatenization of two or more K×K invertible transforms, the POCS projections are calculated in RK space using only the K×K transforms and their inverses, rather than the larger RN space using pseudo inverse transforms. Practical reconstructions are enabled based on, for example, wavelet, subband, or lapped transforms of an entire image. In one embodiment, unequal error protection for multiple description source coding is provided. In particular, given a bit-plane representation of the coefficients in an overcomplete representation of the source, one embodiment of the present invention provides coding the most significant bits with the highest redundancy and the least significant bits with the lowest redundancy. In one embodiment, this is accomplished by varying the quantization stepsize for the different coefficients. Then, the available received quantized coefficients are decoded using a method based on alternating projections onto convex sets.

Abstract translation: 基于凸集（POCS）的方法的投影，用于从从NxK过完全变换接收的量化系数的子集的信号的一致重构。 通过选择一个帧运算符F作为两个或多个KxK可逆变换的并置，POCS投影在RK空间中仅使用KxK变换及其反转而不是使用伪逆变换的较大的RN空间来计算。 基于例如整个图像的小波，子带或重叠变换来实现实际重建。 在一个实施例中，提供了用于多描述源编码的不等差错保护。 特别地，给定源的过完整表示中的系数的位平面表示，本发明的一个实施例提供了具有最高冗余度的最高有效位和具有最低冗余度的最低有效位的编码。 在一个实施例中，这通过改变不同系数的量化步长来实现。 然后，使用基于在凸集上的交替投影的方法对可用的接收量化系数进行解码。

公开(公告)号：US09105271B2

公开(公告)日：2015-08-11

申请号：US12907889

申请日：2010-10-19

Applicant: Sanjeev Mehrotra ,  Wei-Ge Chen

Inventor： Sanjeev Mehrotra ,  Wei-Ge Chen

IPC: G10L19/008 ,  G10L19/24 ,  G10L21/038

CPC classification number: G10L21/038 ,  G10L19/008

Abstract: An audio encoder receives multi-channel audio data comprising a group of plural source channels and performs channel extension coding, which comprises encoding a combined channel for the group and determining plural parameters for representing individual source channels of the group as modified versions of the encoded combined channel. The encoder also performs frequency extension coding. The frequency extension coding can comprise, for example, partitioning frequency bands in the multi-channel audio data into a baseband group and an extended band group, and coding audio coefficients in the extended band group based on audio coefficients in the baseband group. The encoder also can perform other kinds of transforms. An audio decoder performs corresponding decoding and/or additional processing tasks, such as a forward complex transform.

Abstract translation: 音频编码器接收包括一组多个源信道的多声道音频数据，并执行信道扩展编码，其包括对该组的组合信道进行编码，并确定用于表示该组的各个源信道的多个参数，作为编码组合的修改版本 渠道。 编码器还执行频率扩展编码。 频率扩展编码可以包括例如将多声道音频数据中的频带划分成基带组和扩展频带组，并且基于基带组中的音频系数对扩展频带组中的音频系数进行编码。 编码器还可以执行其他类型的转换。 音频解码器执行相应的解码和/或附加处理任务，例如前向复合变换。

公开(公告)号：US08918375B2

公开(公告)日：2014-12-23

申请号：US13222198

申请日：2011-08-31

Applicant: Jin Li ,  Sudipta Sengupta ,  Sanjeev Mehrotra ,  Ran Kalach ,  Paul Adrian Oltean

Inventor： Jin Li ,  Sudipta Sengupta ,  Sanjeev Mehrotra ,  Ran Kalach ,  Paul Adrian Oltean

IPC: G06F17/30 ,  G06F3/06 ,  G06F7/00 ,  G06F11/10

CPC classification number: G06F7/00 ,  G06F3/0608 ,  G06F3/0641 ,  G06F3/0674 ,  G06F11/1004 ,  G06F17/30159

Abstract: The subject disclosure is directed towards partitioning a file into chunks that satisfy a chunk size restriction, such as maximum and minimum chunk sizes, using a sliding window. For file positions within the chunk size restriction, a signature representative of a window fingerprint is compared with a target pattern, with a chunk boundary candidate identified if matched. Other signatures and patterns are then checked to determine a highest ranking signature (corresponding to a lowest numbered Rule) to associate with that chunk boundary candidate, or set an actual boundary if the highest ranked signature is matched. If the maximum chunk size is reached without matching the highest ranked signature, the chunking mechanism regresses to set the boundary based on the candidate with the next highest ranked signature (if no candidates, the boundary is set at the maximum). Also described is setting chunk boundaries based upon pattern detection (e.g., runs of zeros).

Abstract translation: 本发明涉及使用滑动窗口将文件分成满足块大小限制的块，例如最大和最小块大小。 对于块大小限制内的文件位置，将窗口指纹的签名代表与目标模式进行比较，如果匹配则识别出块边界候选。 然后检查其他签名和模式以确定与该块块边界候选者相关联的最高排名签名（对应于最小编号的规则），或者如果最高排名签名匹配则设置实际边界。 如果没有匹配最高排名的签名达到最大块大小，则分块机制基于具有下一个最高排名的签名的候选者（如果没有候选，边界被设置为最大）而退化以设置边界。 还描述了基于模式检测（例如，零的运行）设置块边界。

公开(公告)号：US08426715B2

公开(公告)日：2013-04-23

申请号：US11958331

申请日：2007-12-17

Applicant: John Bregar ,  Andy Precious ,  Sanjeev Mehrotra ,  Serge Smirnov

Inventor： John Bregar ,  Andy Precious ,  Sanjeev Mehrotra ,  Serge Smirnov

IPC: G10H1/40

CPC classification number: G11B20/10527 ,  G11B27/10 ,  G11B2020/10546

Abstract: A low computational power digital audio player achieves beat continuous transitioning between digital audio pieces based on beat metadata, which can be generated via offline processing on a higher computational power computer or via background or idle processing on the digital audio player. The digital audio player produces playlists of beat matching compatible songs based on the metadata, or pick lists of songs that are beat matching compatible with a currently playing song. By facilitating selection of songs with beat matching compatible tempos based on metadata, the beat continuous transitions can be achieved without altering the beat tempo of digital audio pieces, or with simple resampling.

Abstract translation: 低计算能力数字音频播放器可以基于节拍元数据实现数字音频片段之间的节拍连续转换，其可以通过在较高计算能力计算机上的离线处理或经由数字音频播放器上的背景或空闲处理来生成。 数字音频播放器产生基于元数据的匹配兼容歌曲的播放列表，或者选择与当前播放的歌曲兼容的节拍匹配的歌曲列表。 通过促进基于元数据的匹配匹配兼容节奏的歌曲的选择，可以在不改变数字音频片段的节拍速度的情况下或者通过简单的重采样来实现节拍连续转变。

公开(公告)号：US08396114B2

公开(公告)日：2013-03-12

申请号：US12362420

申请日：2009-01-29

Applicant: Chuang Gu ,  Chun-Wei Chan ,  William Chen ,  Stacey Spears ,  Thomas W. Holcomb ,  Chih-Lung Lin ,  Sanjeev Mehrotra

Inventor： Chuang Gu ,  Chun-Wei Chan ,  William Chen ,  Stacey Spears ,  Thomas W. Holcomb ,  Chih-Lung Lin ,  Sanjeev Mehrotra

IPC: H04N7/12 ,  G06K9/36 ,  G06F11/00

CPC classification number: H04N21/4621 ,  H04N19/124 ,  H04N19/14 ,  H04N19/149 ,  H04N19/177 ,  H04N19/61 ,  H04N19/85 ,  H04N21/23439 ,  H04N21/6373 ,  H04N21/8456

Abstract: A video encoding system encodes video streams for multiple bit rate video streaming using an approach that permits the encoded bit rate to vary subject to a peak bit rate and average bit rate constraints for higher quality streams, while a bottom bit rate stream is encoded to achieve a constant chunk rate. The video encoding system also dynamically decides an encoding resolution for segments of the multiple bit rate video streams that varies with video complexity so as to achieve a better visual experience for multiple bit rate streaming.

Abstract translation: 视频编码系统使用允许编码比特率根据高质量流的峰比特率和平均比特率约束而变化的方法对多个比特率视频流的视频流进行编码，而底部比特率流被编码以实现 恒定的块率。 视频编码系统还动态地决定了随着视频复杂度而变化的多个比特率视频流的段的编码分辨率，从而为多个比特率流实现更好的视觉体验。

公开(公告)号：US08386271B2

公开(公告)日：2013-02-26

申请号：US12055223

申请日：2008-03-25

Applicant: Kazuhito Koishida ,  Sanjeev Mehrotra ,  Radhika Jandhyala

Inventor： Kazuhito Koishida ,  Sanjeev Mehrotra ,  Radhika Jandhyala

IPC: G10L21/04

CPC classification number: G10L19/24 ,  G10L19/0017 ,  G10L19/008 ,  G10L19/0212

Abstract: A scalable audio codec encodes an input audio signal as a base layer at a high compression ratio and one or more residual signals as an enhancement layer of a compressed bitstream, which permits a lossless or near lossless reconstruction of the input audio signal at decoding. The scalable audio codec uses perceptual transform coding to encode the base layer. The residual is calculated in a transform domain, which includes a frequency and possibly also multi-channel transform of the input audio. For lossless reconstruction, the frequency and multi-channel transforms are reversible.

Abstract translation: 可扩展音频编解码器将输入音频信号编码为高压缩比的基本层和作为压缩比特流的增强层的一个或多个残留信号，其允许在解码时输入音频信号的无损或近无损重建。 可扩展音频编解码器使用感知变换编码来对基本层进行编码。 在变换域中计算残差，其包括输入音频的频率和可能的多通道变换。 对于无损重建，频率和多通道变换是可逆的。

公开(公告)号：US20120323584A1

公开(公告)日：2012-12-20

申请号：US13595939

申请日：2012-08-27

Applicant: Kazuhito Koishida ,  Sanjeev Mehrotra ,  Chao He ,  Wei-Ge Chen

Inventor： Kazuhito Koishida ,  Sanjeev Mehrotra ,  Chao He ,  Wei-Ge Chen

IPC: G10L19/00

CPC classification number: G10L19/167 ,  G10L19/002 ,  G10L19/008 ,  G10L19/022 ,  G10L19/03 ,  G10L19/038 ,  G10L19/04 ,  G10L19/24

Abstract: An audio decoder provides a combination of decoding components including components implementing base band decoding, spectral peak decoding, frequency extension decoding and channel extension decoding techniques. The audio decoder decodes a compressed bitstream structured by a bitstream syntax scheme to permit the various decoding components to extract the appropriate parameters for their respective decoding technique.

Abstract translation: 音频解码器提供包括实现基带解码，频谱峰值解码，频率扩展解码和信道扩展解码技术的组件的解码组件的组合。 音频解码器解码由比特流语法方案构成的压缩比特流，以允许各种解码组件为它们各自的解码技术提取适当的参数。

公开(公告)号：US08228800B2

公开(公告)日：2012-07-24

申请号：US12364520

申请日：2009-02-03

Applicant: Sanjeev Mehrotra ,  Ying-zong Huang ,  Jin Li ,  Sayandeep Sen ,  Manish Jain ,  Sudipta Sengupta

Inventor： Sanjeev Mehrotra ,  Ying-zong Huang ,  Jin Li ,  Sayandeep Sen ,  Manish Jain ,  Sudipta Sengupta

IPC: G06F11/00

CPC classification number: H04L47/10 ,  H04J3/0682 ,  H04L1/0002 ,  H04L1/0009 ,  H04L1/0015 ,  H04L1/1867 ,  H04L47/115 ,  H04L47/193 ,  H04L47/25 ,  H04L47/283 ,  H04L47/37 ,  Y02D50/10

Abstract: Transmission delays are minimized when packets are transmitted from a source computer over a network to a destination computer. The source computer measures the network's available bandwidth, forms a sequence of output packets from a sequence of data packets, and transmits the output packets over the network to the destination computer, where the transmission rate is ramped up to the measured bandwidth. In conjunction with the transmission, the source computer monitors a transmission delay indicator which it computes using acknowledgement packets it receives from the destination computer. Whenever the indicator specifies that the transmission delay is increasing, the source computer reduces the transmission rate until the indicator specifies that the delay is unchanged. The source computer dynamically decides whether each output packet will be a forward error correction packet or a single data packet, where the decision is based on minimizing the expected transmission delays.

Abstract translation: 当数据包通过网络从源计算机传输到目标计算机时，传输延迟最小化。 源计算机测量网络的可用带宽，形成来自一系列数据分组的输出分组序列，并通过网络将输出分组发送到目标计算机，其中传输速率升高到测量带宽。 结合传输，源计算机监视传输延迟指示符，其使用从目的地计算机接收的确认分组来计算它。 每当指示符指示传输延迟增加时，源计算机降低传输速率，直到指示符指定延迟不变。 源计算机动态地确定每个输出分组是否将是前向纠错分组或单个数据分组，其中决定基于最小化期望的传输延迟。

公开(公告)号：US20120155262A1

公开(公告)日：2012-06-21

申请号：US12970952

申请日：2010-12-17

Applicant: Jin Li ,  Hua Chen ,  Sourabh Jain ,  Sanjeev Mehrotra

Inventor： Jin Li ,  Hua Chen ,  Sourabh Jain ,  Sanjeev Mehrotra

IPC: H04L12/26

CPC classification number: H04L41/0816 ,  H04L43/04 ,  H04L43/0894 ,  H04L47/12 ,  H04L47/193 ,  H04L47/20 ,  H04L47/22 ,  H04L47/2408 ,  H04L47/2416 ,  H04L47/2441 ,  H04L47/263 ,  H04L47/34 ,  Y02D50/10

Abstract: Described are techniques to use adaptive learning to control bandwidth or rate of transmission of a computer on a network. Congestion observations such as packet delay and packet loss are used to compute a congestion signal. The congestion signal is correlated with information about actual congestion on the network, and the transmission rate is adjusted according to the degree of correlation. Transmission rate may not adjust when packet delay or packet loss is not strongly correlated with actual congestion. The congestion signal is adaptively learned. For instance, the relative effects of loss and delay on the congestion signal may change over time. Moreover, an operating congestion level may be minimized by adaptive adjustment.

Abstract translation: 描述了使用自适应学习来控制网络上的计算机的带宽或速率的技术。 拥塞观察，如分组延迟和分组丢失，用于计算拥塞信号。 拥塞信号与网络上的实际拥塞信息相关，传输速率根据相关程度进行调整。 分组延迟或分组丢失与实际拥塞密切相关时，传输速率可能无法调整。 自适应地学习拥塞信号。 例如，丢失和延迟对拥塞信号的相对影响可能会随时间而变化。 此外，可以通过自适应调整来最小化操作拥塞级别。

公开(公告)号：US20120069899A1

公开(公告)日：2012-03-22

申请号：US13306761

申请日：2011-11-29

Applicant: Sanjeev Mehrotra ,  Wei-Ge Chen

Inventor： Sanjeev Mehrotra ,  Wei-Ge Chen

IPC: H04N7/26 ,  G10L19/00

CPC classification number: G10L19/032 ,  H03M7/40 ,  H03M7/4006 ,  H03M7/4093 ,  H03M7/46

Abstract: An encoder performs context-adaptive arithmetic encoding of transform coefficient data. For example, an encoder switches between coding of direct levels of quantized transform coefficient data and run-level coding of run lengths and levels of quantized transform coefficient data. The encoder can determine when to switch between coding modes based on a pre-determined switch point or by counting consecutive coefficients having a predominant value (e.g., zero). A decoder performs corresponding context-adaptive arithmetic decoding.

Abstract translation: 编码器执行变换系数数据的上下文自适应算术编码。 例如，编码器在量化变换系数数据的直接电平的编码和运行长度的运行电平编码和量化的变换系数数据的电平之间切换。 编码器可以基于预定的切换点或通过计算具有主要值（例如，零）的连续系数来确定何时在编码模式之间切换。 解码器执行相应的上下文自适应算术解码。