公开(公告)号：US07072512B2

公开(公告)日：2006-07-04

申请号：US10202223

申请日：2002-07-23

Applicant: Sanjeev Mehrotra

Inventor： Sanjeev Mehrotra

IPC: G06K9/34

CPC classification number: H04N19/12 ,  G06T9/00 ,  H04N19/14 ,  H04N19/17 ,  H04N19/27 ,  H04N19/85

Abstract: For encoding of mixed-content images containing palettized and continuous-tone content, continuous tone content regions in the image are detected and separated. Continuous tone content segmentation classifies pixels as continuous tone content by counting a number of unique pixel values within a pixel neighborhood. Pixels whose count exceeds a threshold are classified as continuous tone content. The technique further scans the image for regions of high continuous tone pixel density. The segmented continuous-tone and palettized content can be encoded separately for efficient compression, and then reassembled at decompression.

Abstract translation: 对于包含调色和连续色调内容的混合内容图像的编码，检测和分离图像中的连续色调内容区域。 连续色调内容分割通过对像素邻域内的唯一像素值进行计数来将像素分类为连续色调内容。 计数超过阈值的像素被分类为连续色调内容。 该技术进一步扫描图像以获得高连续色调像素密度的区域。 分段的连续色调和调色的内容可以单独编码以进行有效的压缩，然后在解压缩时重新组合。

公开(公告)号：US06980695B2

公开(公告)日：2005-12-27

申请号：US10186481

申请日：2002-06-28

Applicant: Sanjeev Mehrotra

Inventor： Sanjeev Mehrotra

IPC: H04N7/26 ,  H04N7/50 ,  G06K9/36

CPC classification number: H04N19/17 ,  H04N19/124 ,  H04N19/152 ,  H04N19/30 ,  H04N19/46 ,  H04N19/61

Abstract: The present invention relates to regulating the quality and/or bitrate of content within mixed content video when the video is compressed subject to a bitrate constraint. For example, a screen capture encoder encodes palletized content within a frame of screen capture video. Subject to an overall bitrate constraint, the encoder then allocates bits for continuous tone content within the frame. By controlling the allocation of bits used to encode the continuous tone content, the encoder regulates bitrate for the continuous tone content. This in turn can allow the encoder to regulate spatial quality and/or overall temporal quality for the video. In one scenario, for screen capture video encoded to a (relatively) constant overall bitrate, the screen capture encoder reduces the bitrate (and quality) of the continuous tone content, instead spending bits to increase the overall frame rate of the video.

Abstract translation: 本发明涉及当视频被比特率约束压缩时，调节混合内容视频内容的质量和/或比特率。 例如，屏幕捕获编码器在屏幕捕获视频的帧内编码码垛内容。 在总体比特率约束的情况下，编码器然后为帧内的连续色调内容分配比特。 通过控制用于编码连续色调内容的位的分配，编码器调节连续色调内容的比特率。 这又可以允许编码器调节视频的空间质量和/或整体时间质量。 在一种情况下，为了编码为（相对）恒定的总比特率的屏幕捕获视频，屏幕捕获编码器降低连续色调内容的比特率（和质量），而是花费比特来增加视频的总体帧速率。

公开(公告)号：US20050249211A1

公开(公告)日：2005-11-10

申请号：US11177258

申请日：2005-07-08

Applicant: Philip Chou ,  Albert Wang ,  Sanjeev Mehrotra ,  Alexander Mohr

Inventor： Philip Chou ,  Albert Wang ,  Sanjeev Mehrotra ,  Alexander Mohr

IPC: H04L1/00 ,  H04L1/06 ,  H04L1/08 ,  G06K5/04 ,  H04J3/24 ,  H04L12/28 ,  H04L12/56

CPC classification number: H04L1/007 ,  H04L1/0009 ,  H04L1/0017 ,  H04L1/06 ,  H04L1/1812 ,  H04L2001/0093

Abstract: A system and method for correcting errors and losses occurring during a receiver-driven layered multicast (RLM) of real-time media over a heterogeneous packet network such as the Internet. This is accomplished by augmenting RLM with one or more layers of error correction information. This allows each receiver to separately optimize the quality of received audio and video information by subscribing to at least one error correction layer. Ideally, each source layer in a RLM would have one or more multicasted error correction data streams (i.e., layers) associated therewith. Each of the error correction layers would contain information that can be used to replace lost packets from the associated source layer. More than one error correction layer is proposed as some of the error correction packets contained in the data stream needed to replace the packets lost in the associated source stream may themselves be lost in transmission. A preferred process for generating the error correction streams involves the use of a unique adaptation of the Forward Error Correction (FEC) techniques. This process encodes the transmission data using a linear transform which adds redundant elements. The redundancy permits losses to be corrected because any of the original data elements can be derived from any of the encoded elements. Thus, as long as enough of the encoded data elements are received so as to equal the number of the original data elements, it is possible to derive all the original elements.

Abstract translation: 一种用于在异构分组网络（例如因特网）下校正在实时媒体的接收机驱动分层多播（RLM）期间发生的错误和损失的系统和方法。 这是通过用一层或多层纠错信息增强RLM来实现的。 这允许每个接收机通过订阅至少一个纠错层来分别优化所接收的音频和视频信息的质量。 理想地，RLM中的每个源层将具有与其相关联的一个或多个多播的纠错数据流（即，层）。 每个纠错层将包含可用于替换相关源层丢失的分组的信息。 提出了多于一个纠错层，因为包含在替换相关源流中丢失的分组所需的数据流中的一些纠错分组本身可能在传输中丢失。 用于产生纠错流的优选过程涉及使用前向纠错（FEC）技术的唯一适配。 该过程使用添加冗余元素的线性变换对传输数据进行编码。 冗余允许修正损失，因为任何原始数据元素可以从任何编码元素导出。 因此，只要接收到足够的编码数据元素以便等于原始数据元素的数量，就有可能导出所有的原始元素。

公开(公告)号：US06584226B1

公开(公告)日：2003-06-24

申请号：US08819587

申请日：1997-03-14

Applicant: Navin Chaddha ,  Albert Wang ,  Sanjeev Mehrotra

Inventor： Navin Chaddha ,  Albert Wang ,  Sanjeev Mehrotra

IPC: G06K946

CPC classification number: G06T9/008 ,  G06T7/223 ,  G06T2207/10016 ,  H04N19/503 ,  H04N19/61 ,  H04N19/91 ,  H04N19/94 ,  H04N19/96

Abstract: Methods and apparatus for processing video data that is divided into frames are presented. In one aspect, this relates to a method for processing video data that is divided into frames. The video data includes a current frame, which has an associated current macroblock, and an adjacent frame, which also has an associated macroblock. The method for processing video data involves obtaining an uncompressed current block that is a part of the current macroblock and an adjacent block that is part of the adjacent macroblock, and calculating a distance between the uncompressed current block and the adjacent block. It is determined whether the distance between the uncompressed current block and the adjacent block is acceptable. If the distance is unacceptable, then the motion between the uncompressed current block and the adjacent block is estimated, and the uncompressed current block is adaptively compressed.

Abstract translation: 提出了分割为帧的视频数据处理方法和装置。 一方面，这涉及一种用于处理被划分成帧的视频数据的方法。 视频数据包括具有相关联的当前宏块的当前帧以及也具有相关宏块的相邻帧。 用于处理视频数据的方法涉及获得作为当前宏块的一部分的未压缩的当前块和作为相邻宏块的一部分的相邻块，以及计算未压缩的当前块与相邻块之间的距离。 确定未压缩的当前块与相邻块之间的距离是否可接受。 如果距离不可接受，则估计未压缩的当前块和相邻块之间的运动，并且自压压缩未压缩的当前块。

公开(公告)号：US06532562B1

公开(公告)日：2003-03-11

申请号：US09316869

申请日：1999-05-21

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

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

IPC: G06F1100

CPC classification number: H04L1/0059 ,  H04L1/0002 ,  H04L1/06 ,  H04L1/08

Abstract: “Correction of errors and losses occurring during a receiver-driven layered multicast (RLM) of real-time media over a heterogeneous packet network such as the Internet is accomplished by augmenting RLM with one or more layers of error correction information. Each receiver separately optimizes the quality of received audio and video information by subscribing to at least one error correction layer. Ideally, each source layer in a RLM would have one or more associated multicasted error correction data streams (i.e., layers). Each error correction layer contains information that can be used to replace lost packets from the associated source layer. More than one error correction layer is proposed as some of the error correction packets contained in the data stream needed to replace the packets lost in the associated source stream may themselves be lost in transmission.”

Abstract translation: 通过异构分组网络（如Internet）的实时媒体接收机驱动分层多播（RLM）中发生的错误和损失的纠正是通过用一层或多层纠错信息增强RLM来实现的，每个接收机分别优化 通过订阅至少一个纠错层，接收的音频和视频信息的质量理想地，RLM中的每个源层将具有一个或多个关联的多播纠错数据流（即，层），每个纠错层包含 可以使用来自相关源层的丢失分组来替代不止一个纠错层，因为包含在数据流中的一些纠错分组被包含在替换相关源流中丢失的分组所需的数据流本身可能在传输中丢失 “。

公开(公告)号：US06470469B1

公开(公告)日：2002-10-22

申请号：US09276842

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: H03M13/35 ,  G06T5/005 ,  G06T5/10 ,  G06T2207/20052 ,  G06T2207/20056 ,  H03M13/03 ,  H03M13/17

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：US09088510B2

公开(公告)日：2015-07-21

申请号：US13718114

申请日：2012-12-18

Applicant: Jin Li ,  Sanjeev Mehrotra ,  Di Xie

Inventor： Jin Li ,  Sanjeev Mehrotra ,  Di Xie

IPC: H04L12/26 ,  H04L12/825 ,  H04L1/00 ,  H04L12/815 ,  H04L12/801 ,  H04L12/813 ,  H04L12/851 ,  H04L12/853

CPC classification number: H04L47/25 ,  H04L1/0002 ,  H04L1/0015 ,  H04L1/0034 ,  H04L43/0882 ,  H04L47/12 ,  H04L47/193 ,  H04L47/20 ,  H04L47/22 ,  H04L47/2408 ,  H04L47/2416 ,  H04L47/2441 ,  H04L47/263 ,  H04L47/34

Abstract: A “Universal Rate Control Mechanism with Parameter Adaptation” (URCMPA) improves real-time communication (RTC) sessions in terms of delay, loss, throughput, and PSNR. The URCMPA automatically learns network characteristics including bottleneck link capacity, inherent queuing delay, inherent packet loss rates, etc., during RTC sessions. The URCMPA uses this information to dynamically adapt rate control parameters in a utility maximization (UM) framework. The URCMPA operates reliable RTC sessions across a wide range and combination of networks near full throughput rates while maintaining low operating congestion levels (e.g., low queuing delay and low packet loss). Examples of networks applicable for use with the URCMPA include, but are not limited to, combinations of mobile broadband (e.g., 3G, 4G, etc.), WiMAX, Wi-Fi hotspots, etc., and physical networks based on cable, fiber, ADSL, etc. The URCMPA can also dynamically adapt operating congestion levels relative to competing TCP flows to maintain fair use of network resources.

Abstract translation: “URCMPA”（“URCMPA”通用速率控制机制）在延迟，丢失，吞吐量和PSNR方面改进了实时通信（RTC）会话。 在RTC会话期间，URCMPA自动学习网络特性，包括瓶颈链路容量，固有排队延迟，固有分组丢失率等。 URCMPA使用此信息在效用最大化（UM）框架中动态调整速率控制参数。 URCMPA在全面吞吐率附近的广泛范围和网络组合中运行可靠的RTC会话，同时保持低的运行拥塞级别（例如，低排队延迟和低分组丢失）。 适用于URCMPA的网络的示例包括但不限于移动宽带（例如，3G，4G等），WiMAX，Wi-Fi热点等的组合以及基于电缆，光纤的物理网络 ，ADSL等。URCMPA还可以动态调整相对于竞争性TCP流的运行拥塞级别，以保持网络资源的合理使用。

公开(公告)号：US08738986B2

公开(公告)日：2014-05-27

申请号：US12718537

申请日：2010-03-05

Applicant: Nelamangal Krishnaswamy Srinivas ,  Nadim Y. Abdo ,  Sanjeev Mehrotra ,  Tong L. Wynn

Inventor： Nelamangal Krishnaswamy Srinivas ,  Nadim Y. Abdo ,  Sanjeev Mehrotra ,  Tong L. Wynn

IPC: H03M13/00

CPC classification number: G06F11/10 ,  H03M13/07

Abstract: In various embodiments, methods and systems are disclosed for integrating a remote presentation protocol with a datagram based transport. In one embodiment, an integrated protocol is configured to support lossless or reduced loss transport based on Retransmission (ARQ) combined with Forward Error Correction (FEC). The protocol involves encoding and decoding of data packets including feedback headers and FEC packets, continuous measurement of RTT, RTO and packet delay, dynamically evaluating loss probability to determine and adjust the ratio of FEC, congestion management based on dynamically detecting increase in packet delay, and fast data transmission rate ramp-up based on detecting a decrease in packet delay.

Abstract translation: 在各种实施例中，公开了用于将远程呈现协议与基于数据报的传输集成的方法和系统。 在一个实施例中，集成协议被配置为支持基于与前向纠错（FEC）组合的重发（ARQ）的无损或减少的丢失传输。 该协议涉及数据包的编码和解码，包括反向报头和FEC分组，RTT连续测量，RTO和分组延迟，动态评估丢失概率，以确定和调整FEC的比例，基于动态检测分组延迟增加的拥塞管理， 并且基于检测到分组延迟的减少，快速数据传输速率上升。

公开(公告)号：US08712783B2

公开(公告)日：2014-04-29

申请号：US13306761

申请日：2011-11-29

Applicant: Sanjeev Mehrotra ,  Wei-Ge Chen

Inventor： Sanjeev Mehrotra ,  Wei-Ge Chen

IPC: G10L19/02 ,  H03M7/46 ,  H03M7/48

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。

公开(公告)号：US08553540B2

公开(公告)日：2013-10-08

申请号：US12762016

申请日：2010-04-16

Applicant: Sanjeev Mehrotra ,  Tong L. Wynn ,  Jin Li ,  Sudipta Sengupta

Inventor： Sanjeev Mehrotra ,  Tong L. Wynn ,  Jin Li ,  Sudipta Sengupta

IPC: H04L12/26

CPC classification number: H04L47/25 ,  H04L47/22 ,  H04L47/2416 ,  H04L47/29 ,  H04L47/30

Abstract: In various embodiments, methods and systems are disclosed for a hybrid rate plus window based congestion protocol that controls the rate of packet transmission into the network and provides low queuing delay, practically zero packet loss, fair allocation of network resources amongst multiple flows, and full link utilization. In one embodiment, a congestion window may be used to control the maximum number of outstanding bits, a transmission rate may be used to control the rate of packets entering the network (packet pacing), a queuing delay based rate update may be used to control queuing delay within tolerated bounds and minimize packet loss, and aggressive ramp-up/graceful back-off may be used to fully utilize the link capacity and additive-increase, multiplicative-decrease (AIMD) rate control may be used to provide fairness amongst multiple flows.

Abstract translation: 在各种实施例中，公开了用于混合速率加上基于窗口的拥塞协议的方法和系统，其控制到网络的分组传输速率并提供低排队延迟，实际上零分组丢失，多个流之间的网络资源的公平分配以及全部 链接利用率。 在一个实施例中，可以使用拥塞窗口来控制未完成比特的最大数量，可以使用传输速率来控制进入网络的分组的速率（分组起搏），基于排队延迟的速率更新可以用于控制 可以利用容忍范围内的排队延迟并尽可能减少分组丢失，并且可以使用积极的提升/优雅退避来充分利用链路容量，并且可以使用加法增加乘法减少（AIMD）速率控制来提供多个 流动。