公开(公告)号：US06571016B1

公开(公告)日：2003-05-27

申请号：US08850957

申请日：1997-05-05

Applicant: Sanjeev Mehrotra ,  Albert S. Wang

Inventor： Sanjeev Mehrotra ,  Albert S. Wang

IPC: G06K936

CPC classification number: H04N19/119 ,  H03M7/3082 ,  H04N19/10 ,  H04N19/115 ,  H04N19/146 ,  H04N19/147 ,  H04N19/19 ,  H04N19/48 ,  H04N19/51 ,  H04N19/593 ,  H04N19/60 ,  H04N19/61 ,  H04N19/63 ,  H04N19/90 ,  H04N19/94 ,  H04N19/96

Abstract: An apparatus and method for encoding video frames is provided. The video frames are divided into blocks for encoding. Encoding of the video blocks utilizes motion detection, motion estimation and adaptive compression, to obtain the desired compression for a particular bit rate. Adaptive compression includes intra compression (without regard to other frames) and inter compression (with regard to other frames). Intra compression, inter compression with motion detection, and inter compression with motion estimation are performed on a block by block basis, as needed. Segmentation is provided to compare encoding of a block with encoding of its sub-blocks, and to select the best block size for encoding.

Abstract translation: 提供了一种用于编码视频帧的装置和方法。 视频帧被分成块进行编码。 视频块的编码利用运动检测，运动估计和自适应压缩，以获得针对特定比特率的期望的压缩。 自适应压缩包括帧内压缩（不考虑其他帧）和帧间压缩（关于其他帧）。 根据需要逐块地执行内部压缩，具有运动检测的帧间压缩和具有运动估计的帧间压缩。 提供分段以比较块的编码与其子块的编码，并选择用于编码的最佳块大小。

公开(公告)号：US08570864B2

公开(公告)日：2013-10-29

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：US08447591B2

公开(公告)日：2013-05-21

申请号：US12130862

申请日：2008-05-30

Applicant: Sanjeev Mehrotra

Inventor： Sanjeev Mehrotra

IPC: G10L19/02

CPC classification number: G10L19/0212 ,  G06F17/147 ,  G10L19/0017 ,  G10L19/022

Abstract: An audio encoder/decoder uses a combination of an overlap windowing transform and block transform that have reversible implementations to provide a reversible, integer-integer form of a lapped transform. The reversible lapped transform permits both lossy and lossless transform domain coding of an audio signal having variable subframe sizes.

Abstract translation: 音频编码器/解码器使用具有可逆实现的重叠窗口变换和块变换的组合来提供重叠变换的可逆整数整数形式。 可逆重叠变换允许具有可变子帧大小的音频信号的有损和无损变换域编码。

公开(公告)号：US08090574B2

公开(公告)日：2012-01-03

申请号：US12907848

申请日：2010-10-19

Applicant: Sanjeev Mehrotra ,  Wei-Ge Chen

Inventor： Sanjeev Mehrotra ,  Wei-Ge Chen

IPC: G10L19/02 ,  H03M7/46 ,  H04B1/66

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：US07953604B2

公开(公告)日：2011-05-31

申请号：US11336618

申请日：2006-01-20

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

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

IPC: G10L19/00

CPC classification number: G10L21/038

Abstract: An audio encoder performs frequency extension coding that comprises determining one or more shape parameters using a displacement vector that corresponds to a displacement of an even number (e.g., an even number of sub-bands between a sub-band in a baseband frequency range and a sub-band in an extended-band frequency range). The shape parameters can be determined on a per-audio-block basis. Restricting a displacement to an even number (in frequency extension coding or in other signal modulation schemes) can improve the quality of reconstructed audio. An audio encoder also can perform frequency extension coding that comprises determining one or more scale parameters at one or more audio blocks, and determining one or more anchor points for interpolating the one or more scale parameters.

Abstract translation: 音频编码器执行频率扩展编码，其包括使用对应于偶数位移的位移矢量来确定一个或多个形状参数（例如，基带频率范围中的子带和偶数个子带之间的偶数个子带） 子带在扩展频带范围内）。 形状参数可以基于每个音频块来确定。 将位移限制为偶数（在频率扩展编码或其他信号调制方案中）可以提高重构音频的质量。 音频编码器还可以执行频率扩展编码，其包括确定一个或多个音频块处的一个或多个缩放参数，以及确定用于内插一个或多个缩放参数的一个或多个定位点。

公开(公告)号：US20110035226A1

公开(公告)日：2011-02-10

申请号：US12907889

申请日：2010-10-19

Applicant: Sanjeev Mehrotra ,  Wei-Ge Chen

Inventor： Sanjeev Mehrotra ,  Wei-Ge Chen

IPC: G10L19/00

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为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：US20110035225A1

公开(公告)日：2011-02-10

申请号：US12907848

申请日：2010-10-19

Applicant: Sanjeev Mehrotra ,  Wei-Ge Chen

Inventor： Sanjeev Mehrotra ,  Wei-Ge Chen

IPC: G10L19/00 ,  G10L21/00

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

Abstract: An audio encoder performs adaptive entropy encoding of audio data. For example, an audio encoder switches between variable dimension vector Huffman coding of direct levels of quantized audio data and run-level coding of run lengths and levels of quantized audio data. The encoder can use, for example, context-based arithmetic coding for coding run lengths and levels. The encoder can determine when to switch between coding modes by counting consecutive coefficients having a predominant value (e.g., zero). An audio decoder performs corresponding adaptive entropy decoding.

Abstract translation: 音频编码器执行音频数据的自适应熵编码。 例如，音频编码器在量化音频数据的直接电平的可变维矢量霍夫曼编码和游程长度的游程级编码以及量化的音频数据的电平之间切换。 编码器可以使用例如用于对运行长度和电平进行编码的基于上下文的算术编码。 编码器可以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。 音频解码器执行相应的自适应熵解码。

公开(公告)号：US07761290B2

公开(公告)日：2010-07-20

申请号：US11764134

申请日：2007-06-15

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

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

IPC: G10L19/02

CPC classification number: G10L19/0208 ,  G10L19/032

Abstract: An audio encoder/decoder performs band partitioning for vector quantization encoding of spectral holes and missing high frequencies that result from quantization when encoding at low bit rates. The encoder/decoder determines a band structure for spectral holes based on two threshold parameters: a minimum hole size threshold and a maximum band size threshold. Spectral holes wider than the minimum hole size threshold are partitioned evenly into bands not exceeding the maximum band size threshold in size. Such hole filling bands are configured up to a preset number of hole filling bands. The bands for missing high frequencies are then configured by dividing the high frequency region into bands having binary-increasing, linearly-increasing or arbitrarily-configured band sizes up to a maximum overall number of bands.

Abstract translation: 音频编码器/解码器对以低比特率进行编码的频谱空间矢量量化编码和由量化产生的缺失高频进行频带划分。 编码器/解码器基于两个阈值参数确定频谱孔的频带结构：最小孔尺寸阈值和最大频带尺寸阈值。 比最小孔尺寸阈值更宽的光谱孔被均匀地分割成不超过最大带尺寸阈值的带。 这样的孔填充带被配置成预定数量的填充孔。 然后通过将高频区域划分成具有二进制增加，线性增加或任意配置的频带大小直到最大总带数的频带来配置用于缺失高频的频带。

公开(公告)号：US20100153822A1

公开(公告)日：2010-06-17

申请号：US12335496

申请日：2008-12-15

Applicant: Cheng Huang ,  Sanjeev Mehrotra ,  Jin Li ,  Xiping Zuo ,  Walid S.I. Ali ,  Regis J. Crinon ,  Ruchir Astavans

Inventor： Cheng Huang ,  Sanjeev Mehrotra ,  Jin Li ,  Xiping Zuo ,  Walid S.I. Ali ,  Regis J. Crinon ,  Ruchir Astavans

IPC: H03M13/05 ,  G06F11/10

CPC classification number: H03M13/134 ,  H03M13/373 ,  H04L1/0057

Abstract: Construction and use of forward error correction codes is provided. A systematic MDS FEC code is obtained having a property wherein any set of contiguous or non-contiguous r packets can be lost during a data transmission of k data packets and r encoded packets and the original k packets can be recovered unambiguously. The systematic MDS FEC code is transformed into a (k+r, k) systematic MDS FEC code that guarantees at least one of the encoded packets is a parity packet. The starting systematic MDS FEC code may be Cauchy-based, and the transformation code derived from the starting Cauchy-based MDS FEC code allows for very efficient initialization, encoding and decoding operations.

Abstract translation: 提供前向纠错码的构造和使用。 获得具有属性的系统MDS FEC代码，其中在k个数据分组和r个编码分组的数据传输期间可能丢失任何一组连续的或不连续的r分组，并且可以明确地恢复原始k个分组。 系统MDS FEC码被变换为（k + r，k）系统MDS FEC码，其保证编码分组中的至少一个是奇偶校验分组。 起始的系统MDS FEC码可以是基于Cauchy的，并且从起始的基于Cauchy的MDS FEC码导出的变换码允许非常有效的初始化，编码和解码操作。

公开(公告)号：US07697514B2

公开(公告)日：2010-04-13

申请号：US11109250

申请日：2005-04-18

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

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

IPC: 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。